Using rtpbin

[canetti]

Hi .
I trying to build video and audio dender and receiver pipelines .

My sender pipeline is :

gst-launch gstrtpbin name=rtpbin videotestsrc ! ffmpegcolorspace !
ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 !
udpsink host=127.0.0.1 port=5000 rtpbin.send_rtcp_src_0 ! udpsink
host=127.0.0.1 port=5001 sync=false async=false udpsrc port=5005 !
rtpbin.recv_rtcp_sink_0 audiotestsrc ! audioconvert ! audioresample !
audio/x-raw-int, rate=11000, channels=1, endianness=1234, depth=16,
signed=true ! faac ! rtpmp4gpay ! rtpbin.send_rtp_sink_1
rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002
rtpbin.send_rtcp_src_1 ! udpsink host=127.0.0.1 port=5003 sync=false
async=false udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

My receiver pipeline is :

gst-launch -v gstrtpbin name=rtpbin udpsrc
caps="application/x-rtp,media=(string)video,clock-rate=(int)21000,encoding-name=(string)H263-1998"
port=5000 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtph263pdepay ! ffdec_h263 !
sdlvideosink udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0
rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false udpsrc
caps = application/x-rtp, media=(string)audio, clock-rate=(int)11025,
encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)1,
streamtype=(string)5, profile-level-id=(string)2, mode=(string)AAC-hbr,
config=(string)1508, sizelength=(string)13, indexlength=(string)3,
indexdeltalength=(string)3, ssrc=(uint)4286476404, payload=(int)96,
clock-base=(uint)1631679159, seqnum-base=(uint)64394" port=5002 !
rtpbin.recv_rtp_sink_1 rtpbin. ! rtpmp4gdepay ! faad ! audio/x-raw-int,
rate=11000, channels=1, endianness=1234, depth=16, signed=true !
sdlaudiosink udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1
rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false

Once both are playing , The receiving side stops and exits the task ,
displaying in the console the following :

ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data
flow
 error.
Additional debug info:
..\..\..\Source\gstreamer\libs\gst\base\gstbasesrc.c(2563):
gst_base_src_loop ()
: /GstPipeline:pipeline0/GstUDPSrc:udpsrc0:
streaming task paused, reason not-linked (-1)
Execution ended after 8810960000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
/GstPipeline:pipeline0/GstUDPSink:udpsink1.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:recv_rtp_src_1_1228256610_96
: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:recv_rtp_src_0_2166813838_96
: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:send_rtcp_src_1: caps =
NULL

/GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:send_rtcp_src_0: caps =
NULL

/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpPtDemux:rtpptdemux1.GstPad:src_96:
 caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpPtDemux:rtpptdemux1.GstPad:sink:
c
aps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpJitterBuffer:rtpjitterbuffer1.GstP
ad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpJitterBuffer:rtpjitterbuffer1.GstP
ad:src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSsrcDemux:rtpssrcdemux1.GstPad:src
_1228256610: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSsrcDemux:rtpssrcdemux1.GstPad:sin
k: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpPtDemux:rtpptdemux0.GstPad:src_96:
 caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpPtDemux:rtpptdemux0.GstPad:sink:
c
aps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpJitterBuffer:rtpjitterbuffer0.GstP
ad:sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpJitterBuffer:rtpjitterbuffer0.GstP
ad:src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:src
_-2128153458: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:sin
k: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSession:rtpsession1.GstPad:send_rt
cp_src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSession:rtpsession1.GstPad:recv_rt
p_src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSession:rtpsession1.GstPad:recv_rt
p_sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSession:rtpsession0.GstPad:send_rt
cp_src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSession:rtpsession0.GstPad:recv_rt
p_src: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin/GstRtpSession:rtpsession0.GstPad:recv_rt
p_sink: caps = NULL
/GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:recv_rtp_sink_1: caps =
NULL

/GstPipeline:pipeline0/GstRtpBin:rtpbin.GstGhostPad:recv_rtp_sink_0: caps =
NULL

/GstPipeline:pipeline0/GstUDPSrc:udpsrc2.GstPad:src: caps = NULL
/GstPipeline:pipeline0/GstUDPSrc:udpsrc0.GstPad:src: caps = NULL
Setting pipeline to NULL ...
Freeing pipeline ...

------------------------------------------------------------------------------------------------------------

What is the reason ?
Are my caps in the receiver side not defined properly ?
What should be done to make this work ?


Thanks ,
Lior .


--
View this message in context: http://gstreamer-devel.966125.n4.nabble.com/Using-rtpbin-tp3724862p3724862.html
Sent from the GStreamer-devel mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freedesktop.org/archives/gstreamer-devel/attachments/20110807/5e3d3bff/attachment.html>