I'm trying to stream audio from a browser to a gstreamer pipeline on a server.
I'm currently using Kurento, and modifying the Hello World example to try to connect an RTP Endpoint to the pipeline -- but am having trouble.
I know the media is getting there because when I swap in a Recording Endpoint, I get a valid recording.
The Kurento Node JS is:
pipeline.create("RtpEndpoint", {}, function(error, rtpEndpoint) { if (error) { console.log("Recorder problem"); return sendError(res, 500, error); } console.log("Creating WebRtcEndpoint"); pipeline.create('WebRtcEndpoint', function(error, webRtcEndpoint) { if (error) { return sendError(res, 500, error); } console.log("Processing sdpOffer at server and generating sdpAnswer"); webRtcEndpoint.processOffer(sdpOffer, function(error, sdpAnswer) { if (error) { webRtcEndpoint.release(); return sendError(res, 500, error); } console.log("Connecting loopback"); webRtcEndpoint.connect(webRtcEndpoint, function(error) { if(error){ webRtcEndpoint.release(); return sendError(res, 500, error); } console.log("Sending sdpAnswer to client"); console.log(sdpAnswer); webRtcEndpoint.connect(rtpEndpoint, function(error) { if(error) { webRtcEndpoint.release(); return sendError(res, 500, error); } rtpEndpoint.generateOffer(function(error, offer) { fs.writeFile('/tmp/test.sdp',offer); console.log("RTP OFFER GENERATED."); }); }); res.type('application/sdp'); res.send(sdpAnswer); }); }); }); }); and my GStreamer pipeline is:
gst-launch-1.0 -vvvv filesrc location=/tmp/test.sdp ! sdpdemux ! decodebin ! autovideosink which returns
Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Got context from element 'autovideosink0-actual-sink-glimage': gst.gl.GLDisplay=context, gst.gl.GLDisplay=(GstGLDisplay)"\(GstGLDisplayX11\)\ gldisplayx11-0"; Setting pipeline to PLAYING ... New clock: GstSystemClock /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstUDPSrc:udpsrc0: timeout = 10000000000 /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstUDPSrc:udpsrc2: timeout = 10000000000 /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession0.GstPad:send_rtcp_src: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstRtpBin:rtpbin0.GstGhostPad:send_rtcp_src_0: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstUDPSink:udpsink0.GstPad:sink: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstRtpBin:rtpbin0.GstGhostPad:send_rtcp_src_0.GstProxyPad:proxypad4: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstRtpBin:rtpbin0/GstRtpSession:rtpsession1.GstPad:send_rtcp_src: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstRtpBin:rtpbin0.GstGhostPad:send_rtcp_src_1: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstUDPSink:udpsink1.GstPad:sink: caps = application/x-rtcp /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0/GstRtpBin:rtpbin0.GstGhostPad:send_rtcp_src_1.GstProxyPad:proxypad7: caps = application/x-rtcp ERROR: from element /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0: Could not read from resource. Additional debug info: gstsdpdemux.c(1213): gst_sdp_demux_handle_message (): /GstPipeline:pipeline0/GstSDPDemux:sdpdemux0: Could not receive any UDP packets for 10.0000 seconds, maybe your firewall is blocking it. Execution ended after 0:00:10.062018001 Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ... It doesnt work on FFPMEG, VLC, etc -- results are similar to "Attempt 5.3" here: https://altanaitelecom.wordpress.com/2015/02/26/continue-streaming-broadcasting-live-video-call-to-non-webrtc-supported-browsers-and-media-players/
I don't think theres a firewall issue as the pipeline and kurento instance are on the same virtual machine (which has no firewall) -- and the recording endpoint works. Is it being linked badly? Is there an easier way?