[00;04mgstrtpmpadepay.c(213) <div style="direction: ltr;">:gst_rtp_mpa_depay_chain:[00m Unexpected payload type 96</div> Ok here's your problem : gst-plugins-base-0.10.6/gst-libs/gst/rtp/gstrtpbuffer.h:  GST_RTP_PAYLOAD_MPA = 14,             /* Audio MPEG 1-3 */ so try with application/x-rtp, media=audio, payload=14, media=(string)audio, clock-rate=(int)90000, encoding-name=(string)MPA' <div> On 6/7/06, Bebjak, Michal <<a href="mailto:michal.bebjak@siemens.com">michal.bebjak@siemens.com</a>> wrote:<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> I tied to start it with --gst-debug-level=5 and I can see this following errors: DEBUG (0x8122190 - 0:00:00.553758000) [00;04m             default[00m([333m 5798[00m) [00;04mgstrtpmpadepay.c(213):gst_rtp_mpa_depay_chain:[00m Unexpected payload type 96 LOG   (0x8122190 - 0:00:00.553790000) [00;01;31;44m     GST_REFCOUNTING[00m([333m 5798[00m) [00;01;31;44mgstminiobject.c(293):gst_mini_object_unref:[00m 0x8119e70 unref 1->0 LOG   (0x8122190 - 0:00:00.553812000) [00;01;42m          GST_BUFFER[00m([333m 5798[00m) [00;01;42mgstbuffer.c(182):gst_buffer_finalize:[00m finalize 0x8119e70 LOG   (0x8122190 - 0:00:00.553834000) [00;01;34m            GST_CAPS[00m([333m 5798[00m) [00;01;34mgstcaps.c(381):gst_caps_unref:[00m 0x8108068 3->2 LOG   (0x8122190 - 0:00:00.553859000) [00;01;35m      GST_SCHEDULING[00m([333m 5798[00m) [00;01;35mgstpad.c(3177):gst_pad_chain:<rtpmpadepay0:sink>[00m called chainfunction &0xb79b0a50 of pad rtpmpadepay0:sink, returned error LOG   (0x8122190 - 0:00:00.553885000) [00;01;31;44m     GST_REFCOUNTING[00m([333m 5798[00m) [00;01;31;44mgstobject.c(412):gst_object_unref:<rtpmpadepay0:sink>[00m 0x80ed1e0 unref 2->1 LOG   (0x8122190 - 0:00:00.553908000) [00;01;31;44m     GST_REFCOUNTING[00m([333m 5798[00m) [00;01;31;44mgstobject.c(412):gst_object_unref:<capsfilter0>[00m 0x81084d8 unref 2->1 LOG   (0x8122190 - 0:00:00.553931000) [00;01;35m      GST_SCHEDULING[00m([333m 5798[00m) [00;01;35mgstpad.c(3177):gst_pad_chain:<capsfilter0:sink>[00m called chainfunction &gst_base_transform_chain of pad capsfilter0:sink, returned error LOG   (0x8122190 - 0:00:00.553956000) [00;01;31;44m     GST_REFCOUNTING[00m([333m 5798[00m) [00;01;31;44mgstobject.c(412):gst_object_unref:<capsfilter0:sink>[00m 0x8107f40 unref 2->1 DEBUG (0x8122190 - 0:00:00.553978000) [00m             basesrc[00m([333m 5798[00m) [00mgstbasesrc.c(1312):gst_base_src_loop:<udpsrc0>[00m pausing task, reason error DEBUG (0x8122190 - 0:00:00.554002000) [00m                task[00m([333m 5798[00m) [00mgsttask.c(438):gst_task_pause:<task0>[00m Pausing task 0x810f9c0 WARN  (0x8122190 - 0:00:00.554038000) [00m             basesrc[00m([333m 5798[00m) [00mgstbasesrc.c(1318):gst_base_src_loop:<udpsrc0>[00m error: Internal data flow error. WARN  (0x8122190 - 0:00:00.554060000) [00m             basesrc[00m([333m 5798[00m) [00mgstbasesrc.c(1318):gst_base_src_loop:<udpsrc0>[00m error: streaming task paused, reason error Do you think problem is that the rtpmpadepay element doesn't accept the "payload" argument?? The rest of the logs looks OK. There are some additional warning messages, but I think that is not the problem. Just to be sure here are the messages: WARN  (0x8052240 - 0:00:00.034834000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstmms.so: undefined symbol: mmsh_connect WARN  (0x8052240 - 0:00:00.040353000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgsttrm.so: undefined symbol: gst_pad_query_peer_duration WARN  (0x8052240 - 0:00:00.040762000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstannodex.so: undefined symbol: g_intern_static_string WARN  (0x8052240 - 0:00:00.041090000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstapetag.so: undefined symbol: gst_type_find_helper_for_buffer WARN  (0x8052240 - 0:00:00.041383000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstauparse.so: undefined symbol: gst_pad_query_peer_duration WARN  (0x8052240 - 0:00:00.041748000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstavi.so: undefined symbol: gst_pad_query_peer_duration WARN  (0x8052240 - 0:00:00.042095000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstdebug.so: undefined symbol: gst_pad_query_peer_duration WARN  (0x8052240 - 0:00:00.042571000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgsticydemux.so: undefined symbol: gst_type_find_helper_for_buffer WARN  (0x8052240 - 0:00:00.042983000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstid3demux.so: undefined symbol: gst_tag_from_id3_user_tag WARN  (0x8052240 - 0:00:00.048695000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstossaudio.so: undefined symbol: g_intern_static_string WARN  (0x8052240 - 0:00:00.050635000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstshout2.so: undefined symbol: gst_base_sink_set_sync WARN  (0x8052240 - 0:00:00.051117000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstwavparse.so: undefined symbol: gst_adapter_new WARN  (0x8052240 - 0:00:00.053315000)   GST_PLUGIN_LOADING( 5834) gstplugin.c(414):gst_plugin_load_file: module_open failed: /usr/lib/gstreamer-0.10/libgstswfdec.so: undefined symbol: swfdec_decoder_get_version Best regards, Michal -----Original Message----- From: Antoine Tremblay [mailto:<a href="mailto:hexa00@gmail.com">hexa00@gmail.com</a>] Sent: Tue 6/6/2006 11:54 PM To: Bebjak, Michal Cc: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Subject: Re: [gst-devel] problem sending audio/mpeg over RTP hehe now that I think of it it should work with the default ....but still it's a good thing to set it just to be sure... try to run it with --gst-debug-level=5 or any level and check what's happening juste before the error... On 6/6/06, Antoine Tremblay <<a href="mailto:hexa00@gmail.com"> hexa00@gmail.com</a>> wrote: > > Well you're missing some args in udpsink /udpsrc > > Should be like udpsink host=localhost port=7777 > > udpsrc port=7777 > > the rest seems ok at 1st glance... > > Regards > > Antoine > > > > On 6/6/06, Bebjak, Michal <<a href="mailto:michal.bebjak@siemens.com">michal.bebjak@siemens.com</a>> wrote: > > > > > > Hi, > > > > I'm developing an client-server aplication which should use GStreamer. I > > want to endcode the audio into MP3 and send it oved RTP to the client. I > > first tried to run this following commands: > > > > server terminal:  gst-launch-0.10 -v audiotestsrc ! lame ! rtpmpapay ! > > udpsink > > client terminal:  gst-launch-0.10 -v udpsrc ! 'application/x-rtp, > > media=audio, payload=96, media=(string)audio, clock-rate=(int)90000, > > encoding-name=(string)MPA' ! > >                   rtpmpadepay ! > > 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' ! mad ! > > audioconvert ! volume volume=0.2 ! autoaudiosink > > > > > > They both work independently but when I try to send the audio from > > server to client the client crashes. The terminal output is: > > > > > > $ gst-launch-0.10 -v udpsrc ! 'application/x-rtp, media=audio, > > payload=96, media=(string)audio, clock-rate=(int)90000, > > encoding-name=(string)MPA' ! > > rtpmpadepay ! 'audio/mpeg,mpegversion=1,layer=3,channels=1,rate=44100' ! > > mad ! audioconvert ! volume volume=0.2 ! autoaudiosink > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > /pipeline0/capsfilter0.sink: caps = application/x-rtp, > > media=(string)audio, payload=(int)96, clock-rate=(int)90000, > > encoding-name=(string)MPA > > /pipeline0/capsfilter0.src: caps = application/x-rtp, > > media=(string)audio, payload=(int)96, clock-rate=(int)90000, > > encoding-name=(string)MPA > > /pipeline0/rtpmpadepay0.sink: caps = application/x-rtp, > > media=(string)audio, payload=(int)96, clock-rate=(int)90000, > > encoding-name=(string)MPA > > ERROR: from element /pipeline0/udpsrc0: Internal data flow error. > > Additional debug info: > > gstbasesrc.c(1318): gst_base_src_loop (): /pipeline0/udpsrc0: > > streaming task paused, reason error > > Execution ended after 1513470000 ns. > > Setting pipeline to PAUSED ... > > Setting pipeline to READY ... > > /pipeline0/rtpmpadepay0.sink: caps = NULL > > /pipeline0/capsfilter0.sink: caps = NULL > > /pipeline0/capsfilter0.src: caps = NULL > > Setting pipeline to NULL ... > > FREEING pipeline ... > > > > Can someone please help me to make it work?? Thanks a lot!! > > > > Michael > > > > > > _______________________________________________ > > gstreamer-devel mailing list > > <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> > > <a href="https://lists.sourceforge.net/lists/listinfo/gstreamer-devel"> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel</a> > > > > </blockquote></div>