Thanks its working.I have used the recv_rtp_src_m (where m is the session identifier) and used the same function GST_PAD_NAME (new_pad) to get this name. passed the depayloaders in an array in g_signal_connect.We can do other way also checking the caps from pad and from there we can search payload type or media type . <div class="gmail_quote">On Mon, Aug 1, 2011 at 12:53 AM, Andy_Milestone <<a href="mailto:andy87.milestone@googlemail.com">andy87.milestone@googlemail.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"> Well, I had the same problem, because I am working on softphone with video- and audio support. I am not sure If there is an "intelligent" solution for this problem, but I checked the payload-type, which is given in this line:<div class="im"> > g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad)); </div> If its payloadtype 0 then I connect the rtpbin with the audio-depay and if its 96 I connect it to the video-depay. In my case the rtpbin is defined as a pointer in a global structure, therefore I can decide in the callback which depayloader should be used. Do want to realize undirectional or bidirectional streaming? Andreas Am 01.08.2011 06:15, schrieb arpi d: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div></div><div class="h5"> I am writing client code which receives audio+video. When I am trying to connect rtpbin and depayloader how to know which session is this? Actually,I am writing the code in C and am trying to use g_signal_connect. So that I can use g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), videodepay); g_signal_connect (rtpbin, "pad-added", G_CALLBACK (pad_added_cb), audiodepay); and the pad_added_cb handler is: |/* will be called when rtpbin has validated a payload that we can depayload */ |static void |pad_added_cb (GstElement * rtpbin, GstPad * new_pad, GstElement * depay) { GstPad *sinkpad; GstPadLinkReturn lres; g_print ("new payload on pad: %s\n", GST_PAD_NAME (new_pad)); sinkpad = gst_element_get_static_pad (depay,"sink"); g_assert (sinkpad); lres = gst_pad_link (new_pad, sinkpad); g_assert (lres == GST_PAD_LINK_OK); gst_object_unref (sinkpad); }| my client pipeline is: VIDEO_DEC="rtph264depay ! ffdec_h264" AUDIO_DEC="rtpmp4gdepay ! faad" VIDEO_SINK="ffmpegcolorspace ! autovideosink" AUDIO_SINK="autoaudiosink" DEST=127.0.0.1 LATENCY=100 gst-launch -v gstrtpbin name=rtpbin \ udpsrc caps=$VIDEO_CAPS port=5000 ! rtpbin.recv_rtp_sink_0 \ rtpbin. ! $VIDEO_DEC ! $VIDEO_SINK \ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ rtpbin.send_rtcp_src_0 ! udpsink port=5005 host=$DEST sync=false async=false udpsrc caps=$AUDIO_CAPS port=5002 ! rtpbin.recv_rtp_sink_1 \ rtpbin. ! $AUDIO_DEC ! $AUDIO_SINK \ udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=$DEST sync=false async=false Same thing as above pipeline in C language I am writing. Please reply. If you know other way please do let me know. Thaks in advance. </div></div> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org" target="_blank">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div>