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<title>rtprtxreceive: GStreamer Good Plugins 1.0 Plugins Reference Manual</title>
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Top |
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Description |
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Object Hierarchy |
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Properties
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rtprtxreceive
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rtprtxreceive
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Properties
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guint
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num-rtx-requests
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Read
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guint
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num-rtx-packets
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Read
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guint
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num-rtx-assoc-packets
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Read
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GstStructure *
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payload-type-map
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Read / Write
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Types and Values
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struct
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GstRtpRtxReceive
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Object Hierarchy
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GObject
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╰── GInitiallyUnowned
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╰── GstObject
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╰── GstElement
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╰── GstRtpRtxReceive
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Description
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rtprtxreceive listens to the retransmission events from the
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downstream rtpjitterbuffer and remembers the SSRC (ssrc1) of the stream and
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the sequence number that was requested. When it receives a packet with
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a sequence number equal to one of the ones stored and with a different SSRC,
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it identifies the new SSRC (ssrc2) as the retransmission stream of ssrc1.
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From this point on, it replaces ssrc2 with ssrc1 in all packets of the
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ssrc2 stream and flags them as retransmissions, so that rtpjitterbuffer
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can reconstruct the original stream.
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This algorithm is implemented as specified in RFC 4588.
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This element is meant to be used with rtprtxsend on the sender side.
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See GstRtpRtxSend
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Below you can see some examples that illustrate how rtprtxreceive and
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rtprtxsend fit among the other rtp elements and how they work internally.
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Normally, hoewever, you should avoid using such pipelines and use
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rtpbin instead, with its “request-aux-sender” and
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“request-aux-receiver” signals. See GstRtpBin.
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Example pipelines
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gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! \
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rtprtxsend payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \
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rtpsession.send_rtp_sink \
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rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \
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udpsink host="127.0.0.1" port=5000 \
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udpsrc port=5001 ! rtpsession.recv_rtcp_sink \
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rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 \
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sync=false async=false
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Send audio stream through port 5000 (5001 and 5002 are just the rtcp
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link with the receiver)
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gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
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rtpsession.recv_rtp_sink \
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rtpsession.recv_rtp_src ! \
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rtprtxreceive payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \
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rtpssrcdemux ! rtpjitterbuffer do-retransmission=true ! \
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rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
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rtpsession.send_rtcp_src ! \
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udpsink host="127.0.0.1" port=5001 sync=false async=false \
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udpsrc port=5002 ! rtpsession.recv_rtcp_sink
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Receive audio stream from port 5000 (5001 and 5002 are just the rtcp
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link with the sender)
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In this example we can see a simple streaming of an OPUS stream with some
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of the packets being artificially dropped by the identity element.
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Thanks to retransmission, you should still hear a clear sound when setting
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drop-probability to something greater than 0.
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Internally, the rtpjitterbuffer will generate a custom upstream event,
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GstRTPRetransmissionRequest, when it detects that one packet is missing.
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Then this request is translated to a FB NACK in the rtcp link by rtpsession.
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Finally the rtpsession of the sender side will re-convert it in a
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GstRTPRetransmissionRequest that will be handled by rtprtxsend. rtprtxsend
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will then re-send the missing packet with a new srrc and a different payload
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type (here, 97), but with the same original sequence number. On the receiver
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side, rtprtxreceive will associate this new stream with the original and
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forward the retransmission packets to rtpjitterbuffer with the original
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ssrc and payload type.
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gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=97 seqnum-offset=1 ! \
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rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
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funnel name=f ! rtpsession.send_rtp_sink \
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audiotestsrc freq=660.0 is-live=true ! opusenc ! \
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rtpopuspay pt=97 seqnum-offset=100 ! \
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rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
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f. \
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rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \
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udpsink host="127.0.0.1" port=5000 \
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udpsrc port=5001 ! rtpsession.recv_rtcp_sink \
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rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 \
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sync=false async=false
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Send two audio streams to port 5000.
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gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97" ! \
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rtpsession.recv_rtp_sink \
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rtpsession.recv_rtp_src ! \
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rtprtxreceive payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
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rtpssrcdemux name=demux \
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demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \
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opusdec ! audioconvert ! autoaudiosink \
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demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \
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opusdec ! audioconvert ! autoaudiosink \
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udpsrc port=5002 ! rtpsession.recv_rtcp_sink \
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rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \
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sync=false async=false
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Receive two audio streams from port 5000.
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In this example we are streaming two streams of the same type through the
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same port. They, however, are using a different SSRC (ssrc is randomly
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generated on each payloader - rtpopuspay in this example), so they can be
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identified and demultiplexed by rtpssrcdemux on the receiver side. This is
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an example of SSRC-multiplexing.
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It is important here to use a different starting sequence number
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(seqnum-offset), since this is the only means of identification that
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rtprtxreceive uses the very first time to identify retransmission streams.
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It is an error, according to RFC4588 to have two retransmission requests for
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packets belonging to two different streams but with the same sequence number.
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Note that the default seqnum-offset value (-1, which means random) would
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work just fine, but it is overriden here for illustration purposes.
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Synopsis
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Element Information
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plugin
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rtpmanager
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author
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Julien Isorce <julien.isorce@collabora.co.uk>
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class
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Codec
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Element Pads
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direction
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presence
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application/x-rtp
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name
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direction
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source
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presence
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application/x-rtp
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Functions
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Types and Values
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struct GstRtpRtxReceive
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struct GstRtpRtxReceive;
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Property Details
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The “num-rtx-requests” property
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“num-rtx-requests” guint
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Number of retransmission events received.
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Flags: Read
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Default value: 0
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The “num-rtx-packets” property
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“num-rtx-packets” guint
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Number of retransmission packets received.
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Flags: Read
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Default value: 0
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The “num-rtx-assoc-packets” property
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“num-rtx-assoc-packets” guint
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Number of retransmission packets correctly associated with retransmission requests.
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Flags: Read
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Default value: 0
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The “payload-type-map” property
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“payload-type-map” GstStructure *
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Map of original payload types to their retransmission payload types.
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Flags: Read / Write
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See Also
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rtprtxsend, rtpsession, rtpjitterbuffer
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Generated by GTK-Doc V1.27
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