RTP dynamic bitrate depending on network condition

[chakra]

I am using GStreamer WebRTCbin in send only mode to send videos to browser

On the Webrtc TRANSCEIVER I have enabled:
       / trans->direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
        g_object_set(trans, "fec-type", GST_WEBRTC_FEC_TYPE_ULP_RED, NULL);
        g_object_set(trans, "do-nack", TRUE, NULL);
/
But in the SDP generated as offer from GStreamer to Browser I could see
only,

               / "type":"offer",
		"sdp":"
		v=0\r\no=- 8618982146125398909 0 IN IP4 0.0.0.0\r\n
		s=-\r\n
		t=0 0\r\n
		a=ice-options:trickle\r\n
		a=group:BUNDLE video0\r\n
		m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100\r\n
		c=IN IP4 0.0.0.0\r\n
		a=setup:actpass\r\n
		a=ice-ufrag:UPTOWGnTMt68LX4oLCOYv6xVOFEEOzpM\r\n
		a=ice-pwd:lkvHifShkaD2rzwQZMBENmlsCdmO2Lhz\r\n
		a=rtcp-mux\r\n
		a=rtcp-rsize\r\n
		a=sendonly\r\n
		a=rtpmap:96 H264/90000\r\n
		a=rtcp-fb:96 nack\r\n
		a=rtcp-fb:96 nack pli\r\n
		a=rtpmap:97 red/90000\r\n
		a=rtpmap:98 ulpfec/90000\r\n
		a=rtpmap:99 rtx/90000\r\n
		a=fmtp:99 apt=97\r\n
		a=rtpmap:100 rtx/90000\r\n
		a=fmtp:100 apt=96\r\n/

So there is no rtcp-fb of payload type REMB

Is the congestion control/REMB implemeted by GStreamer? I see a issue,
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1009 where
in one of the comments its mentioned - "don't implement any congestion
control / automatic bitrate adaptation yet" 

So is it that GStreamer does not support REMB yet and need to depend on
stats like packets-received, packets-lost to get information close to it?




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