rtpsession not sending NACKs?

[marcin at saepia.net]

Hello,

I am using GStreamer 1.8.2 and I am trying to figure out how to use
rtpsession with retransmission.

Current examples in the docs are invalid (
https://bugzilla.gnome.org/show_bug.cgi?id=771383).

I have made the following pipelines:

gst-launch-1.0 -v -m rtpsession name=rtpsession \
        audiotestsrc ! audioconvert ! audioresample ! opusenc ! rtpopuspay
! rtprtxsend payload-type-map=map,96=97 ! \
            identity drop-probability=0.1 ! rtpsession.send_rtp_sink \
            rtpsession.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \
        udpsrc port=5001 ! rtpsession.recv_rtcp_sink \
        rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002
sync=false async=false



gst-launch-1.0 -v -m rtpsession name=rtpsession \
        udpsrc port=5000 caps="application/x-rtp, media=(string)audio,
clock-rate=(int)48000, encoding-name=(string)OPUS, payload=(int)96" ! \
            rtpsession.recv_rtp_sink \
            rtpsession.recv_rtp_src ! rtprtxreceive
payload-type-map=map,96=97 ! rtpjitterbuffer do-retransmission=true !
rtpopusdepay ! \
            opusdec ! audioconvert ! audioresample ! autoaudiosink \
        rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001
sync=false async=false \
        udpsrc port=5002 ! rtpsession.recv_rtcp_sink



But rtpsession refuses to send NACKs. GST_DEBUG log does not reveal, why.
I've read the source code and there are many possible reasons for doing so.
I've checked with Wireshark and indeed RTCP packets with retransmission
request are not sent. Retransmission events are properly emitted from the
RTP jitter buffer.

What am I doing wrong?

Marcin
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