I am implementing a simple RTP streaming application based on the C-API with rtpbin element:
appsrc → audioresample → audioconvert → rtpL16pay (SSRC=9) → queue → rtpbin
If the pad-added signal for template send_rtp_src_%u is received an udpsink element is linked on demand to the pad of the rtpbin.
I only set the adress and port properties of the udpsink element and “is-live” = true and “do-timestamp = true” for the appsrc element.
This works as expected and the RTP stream can be received with:
gst-launch-1.0 -v ^
rtpbin name=rtpbin ^
udpsrc port=5005 caps="application/x-rtp,media=audio,encoding-name=L16,clock-rate=48000,payload=96,channels=1" ! ^
rtpbin.recv_rtp_sink_0 ^
rtpbin.recv_rtp_src_0_9_96 ! rtpL16depay ! fakesink
But there are cyclically warnings and errors emitted by rtpsession0 for my stream with SSRC 9:
running time not set, can not create SR for SSRC 9 (rtp_source_get_new_sr)
source 00000009 can not generate RTCP (generate_rtcp)
generated empty RTCP messages for all the sources (rtp_session_on_timeout)
Are these false-positives or did I forget to set any properties or do I have to handle any signal callbacks?