I’m trying to use the webrtcbin (https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad/html/gst-plugins-bad-plugins-webrtcbin.html) GStreamer element for communicating with MediaSoup.
Without entering into the GStreamer details, the element expects to exchange the SDP info with a webrtc peer (a web example here https://github.com/centricular/gstwebrtc-demos/blob/master/sendrecv/js/webrtc.js).
I’m trying to reconstruct the SDP message from the JSON returned from mediasoup server. The communication starts, but I get these errors:
RTC::Transport::ReceiveRtpPacket() | no suitable Producer for received RTP packet [ssrc:824347444, payloadType:96]
ssrc value is the same sent into the
rtpParameters when creating the producer. I don’t understand if the
rid values can be used if the
ssrc is not found into
Anyway, what I need to know is how
medasoup can interoperate with a client without using the