rtpsession not sending NACKs?

[saepia]

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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[Arjen Veenhuizen]

I think you ran into bug 758719 which I reported some time ago. Apparently the problem is still not fixed... :(