rtprtxqueue

rtprtxqueue maintains a queue of transmitted RTP packets, up to a configurable limit (see max-size-time, max-size-packets), and retransmits them upon request from the downstream rtpsession (GstRTPRetransmissionRequest event).

This element is similar to rtprtxsend, but it has differences:

• Retransmission from rtprtxqueue is not RFC 4588 compliant. The retransmitted packets have the same ssrc and payload type as the original stream.
• As a side-effect of the above, rtprtxqueue does not require the use of rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to reconstruct the stream.
• Retransmission from rtprtxqueue happens as soon as the next regular flow packet is chained, while rtprtxsend retransmits as soon as the retransmission event is received, using a helper thread.
• rtprtxqueue can be used with rtpbin without the need of hooking to its request-aux-sender signal, which means it can be used with rtpbin using gst-launch.

See also rtprtxsend, rtprtxreceive

Example pipelines

 gst-launch-1.0 rtpbin name=b rtp-profile=avpf \
    audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \
    b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \
    udpsrc port=5001 ! b.recv_rtcp_sink_0 \
    b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false

Sender pipeline

 gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \
    udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
        b.recv_rtp_sink_0 \
    b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
    udpsrc port=5002 ! b.recv_rtcp_sink_0 \
    b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false

Receiver pipeline