<HTML><BODY>Hello, all. I have a rtsp source and I need to restream it through my rtsp server. The rtsp source can stream audio/video and sometimes only video. I can create either audio/video pipeline or only video pipeline. But I can't create working pipeline for these both cases. I connect callback to 'pad-added' event and then I link the first video element and the first audio element (if audio exists) to rtspsrc element in 'pad-added' callback. I create/add/link audio elements in 'pad-added' callback but the rtsp client has no audio in this case. Please tell me what is wrong. This pipeline works well with audio-video: Gst.parse_launch( '( rtspsrc location="rtsp://admin:admin@192.168.7.217" ' 'latency=0 ' 'timeout=5000000 ' 'name=rtsp_src ' 'rtsp_src. ' '! queue' '! rtph264depay ' '! rtph264pay ' 'name=pay0 ' 'rtsp_src. ' '! queue' '! rtppcmudepay ' '! rtppcmupay ' 'name=pay1 )' ) OS: gentoo, gstreamer: version 1.6.3, gst-rtsp-server: 1.6.2, python3 Code: #!/usr/bin/env python3 """RTSP restreamer based on GStreamer.""" import gi gi.require_version('Gst', '1.0') gi.require_version('GstRtspServer', '1.0') from gi.repository import Gst, GstRtspServer, GObject loop = GObject.MainLoop() GObject.threads_init() Gst.init(None) class AVPipeline(Gst.Pipeline): def __init__(self): Gst.Pipeline.__init__(self) # rtsp source rtspsrc = Gst.ElementFactory.make('rtspsrc', None) rtspsrc.set_property('location', 'rtsp://admin:admin@192.168.7.217') rtspsrc.set_property('latency', 500) rtspsrc.set_property('timeout', 2000000) self.add(rtspsrc) self.link(rtspsrc) rtspsrc.connect('pad-added', self.on_pad_added) # video vqueue = Gst.ElementFactory.make('queue', None) rtph264depay = Gst.ElementFactory.make('rtph264depay', None) rtph264pay = Gst.ElementFactory.make('rtph264pay', None) rtph264pay.set_property('name', 'pay0') rtph264pay.set_property('pt', 96) self.add(vqueue) self.add(rtph264depay) self.add(rtph264pay) vqueue.link(rtph264depay) rtph264depay.link(rtph264pay) self._tolink_video_elem = vqueue def on_pad_added(self, element, pad): string = pad.query_caps(None).to_string() if string.startswith('application/x-rtp'): if 'media=(string)video' in string: pad.link(self._tolink_video_elem.get_static_pad('sink')) print('Video connected') elif 'media=(string)audio' in string: # create audio # Client doesn't get audio when I add audio elements in this point #audio aqueue = Gst.ElementFactory.make('queue', None) rtppcmudepay = Gst.ElementFactory.make('rtppcmudepay', None) rtppcmupay = Gst.ElementFactory.make('rtppcmupay', None) rtppcmupay.set_property('name', 'pay1') self.add(aqueue) self.add(rtppcmudepay) self.add(rtppcmupay) aqueue.link(rtppcmudepay) rtppcmudepay.link(rtppcmupay) for elem in (aqueue, rtppcmudepay, rtppcmupay): elem.sync_state_with_parent() pad.link(aqueue.get_static_pad('sink')) print('Audio connected') class MyRTSPMediaFactory(GstRtspServer.RTSPMediaFactory): LATENCY = 10000 def __init__(self): GstRtspServer.RTSPMediaFactory.__init__(self) self.set_shared(True) self.set_property('latency', self.LATENCY) self.set_transport_mode(GstRtspServer.RTSPTransportMode.PLAY) def do_create_element(self, url): return AVPipeline() class Restreamer(object): def __init__(self, host, port): self._server = GstRtspServer.RTSPServer() self._server.set_address(host) self._server.set_service(str(port)) mount_points = self._server.get_mount_points() factory = MyRTSPMediaFactory() mount_points.add_factory('/test', factory) self._server.attach(None) def main(): server = Restreamer('0.0.0.0', 9999) print('Started %s:%s' % (server._server.get_address(), server._server.get_service())) loop.run() if __name__ == '__main__': main() </BODY></HTML>