rtpjitterbuffer

This element reorders and removes duplicate RTP packets as they are received from a network source.

The element needs the clock-rate of the RTP payload in order to estimate the delay. This information is obtained either from the caps on the sink pad or, when no caps are present, from the request-pt-map signal. To clear the previous pt-map use the clear-pt-map signal.

The rtpjitterbuffer will wait for missing packets up to a configurable time limit using the latency property. Packets arriving too late are considered to be lost packets. If the do-lost property is set, lost packets will result in a custom serialized downstream event of name GstRTPPacketLost. The lost packet events are usually used by a depayloader or other element to create concealment data or some other logic to gracefully handle the missing packets.

The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incoming buffer and the rtptime inside the RTP packet to create a PTS on the outgoing buffer.

The jitterbuffer can also be configured to send early retransmission events upstream by setting the do-retransmission property. In this mode, the jitterbuffer tries to estimate when a packet should arrive and sends a custom upstream event named GstRTPRetransmissionRequest when the packet is considered late. The initial expected packet arrival time is calculated as follows:

  • If seqnum N arrived at time T, seqnum N+1 is expected to arrive at T + packet-spacing + rtx-delay. The packet spacing is calculated from the DTS (or PTS is no DTS) of two consecutive RTP packets with different rtptime.

  • If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm, seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any previously scheduled timeout is overwritten.

  • If seqnum N arrived, all seqnum older than N - rtx-delay-reorder are considered late immediately. This is to request fast feedback for abnormally reorder packets before any of the previous timeouts is triggered.

A late packet triggers the GstRTPRetransmissionRequest custom upstream event. After the initial timeout expires and the retransmission event is sent, the timeout is scheduled for T + rtx-retry-timeout. If the missing packet did not arrive after rtx-retry-timeout, a new GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled again for T + rtx-retry-timeout. This repeats until rtx-retry-period elapsed, at which point no further retransmission requests are sent and the regular logic is performed to schedule a lost packet as discussed above.

This element acts as a live element and so adds latency to the pipeline.

This element will automatically be used inside rtpbin.

Example pipelines

 gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink

Connect to a streaming server and decode the MPEG video. The jitterbuffer is inserted into the pipeline to smooth out network jitter and to reorder the out-of-order RTP packets.

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──rtpjitterbuffer

Factory details

Authors: – Philippe Kalaf , Wim Taymans

Classification:Filter/Network/RTP

Rank – none

Plugin – gstrtpmanager

Package – GStreamer Good Plug-ins git

Pad Templates

sink

application/x-rtp:

Presencealways

Directionsink


sink_rtcp

application/x-rtcp:

Presencerequest

Directionsink


src

application/x-rtp:

Presencealways

Directionsrc


Signals

clear-pt-map

clear_pt_map_callback (GstElement * buffer,
                       gpointer udata)
def clear_pt_map_callback (buffer, udata):
    #python callback for the 'clear-pt-map' signal
function clear_pt_map_callback(buffer: GstElement * buffer, udata: gpointer udata): {
    // javascript callback for the 'clear-pt-map' signal
}

Invalidate the clock-rate as obtained with the request-pt-map signal.

Parameters:

buffer

the object which received the signal

udata
No description available

handle-sync

handle_sync_callback (GstElement * buffer,
                      GstStructure * struct,
                      gpointer udata)
def handle_sync_callback (buffer, struct, udata):
    #python callback for the 'handle-sync' signal
function handle_sync_callback(buffer: GstElement * buffer, struct: GstStructure * struct, udata: gpointer udata): {
    // javascript callback for the 'handle-sync' signal
}

Be notified of new sync values.

Parameters:

buffer

the object which received the signal

struct

a GstStructure containing sync values.

udata
No description available

on-npt-stop

on_npt_stop_callback (GstElement * buffer,
                      gpointer udata)
def on_npt_stop_callback (buffer, udata):
    #python callback for the 'on-npt-stop' signal
function on_npt_stop_callback(buffer: GstElement * buffer, udata: gpointer udata): {
    // javascript callback for the 'on-npt-stop' signal
}

Signal that the jitterbufer has pushed the RTP packet that corresponds to the npt-stop position.

Parameters:

buffer

the object which received the signal

udata
No description available

request-pt-map

GstCaps *
request_pt_map_callback (GstElement * buffer,
                         guint pt,
                         gpointer udata)
def request_pt_map_callback (buffer, pt, udata):
    #python callback for the 'request-pt-map' signal
function request_pt_map_callback(buffer: GstElement * buffer, pt: guint pt, udata: gpointer udata): {
    // javascript callback for the 'request-pt-map' signal
}

Request the payload type as GstCaps for pt.

Parameters:

buffer

the object which received the signal

pt

the pt

udata
No description available
Returns ( GstCaps * ) –
No description available

set-active

guint64
set_active_callback (GstElement * buffer,
                     gboolean param_1,
                     guint64 param_2,
                     gpointer udata)
def set_active_callback (buffer, param_1, param_2, udata):
    #python callback for the 'set-active' signal
function set_active_callback(buffer: GstElement * buffer, param_1: gboolean param_1, param_2: guint64 param_2, udata: gpointer udata): {
    // javascript callback for the 'set-active' signal
}

Start pushing out packets with the given base time. This signal is only useful in buffering mode.

Parameters:

buffer

the object which received the signal

param_1
No description available
param_2
No description available
udata
No description available
Returns ( guint64 ) –

the time of the last pushed packet.


Properties

do-lost

“do-lost” gboolean

Send out a GstRTPPacketLost event downstream when a packet is considered lost.

Flags : Read / Write

Default value : false


do-retransmission

“do-retransmission” gboolean

Send out a GstRTPRetransmission event upstream when a packet is considered late and should be retransmitted.

Flags : Read / Write

Default value : false


drop-on-latency

“drop-on-latency” gboolean

Drop oldest buffers when the queue is completely filled.

Flags : Read / Write

Default value : false


faststart-min-packets

“faststart-min-packets” guint

The number of consecutive packets needed to start (set to 0 to disable faststart. The jitterbuffer will by default start after the latency has elapsed)

Flags : Read / Write

Default value : 0


latency

“latency” guint

The maximum latency of the jitterbuffer. Packets will be kept in the buffer for at most this time.

Flags : Read / Write

Default value : 200


max-dropout-time

“max-dropout-time” guint

The maximum time (milliseconds) of missing packets tolerated.

Flags : Read / Write

Default value : 60000


max-misorder-time

“max-misorder-time” guint

The maximum time (milliseconds) of misordered packets tolerated.

Flags : Read / Write

Default value : 2000


max-rtcp-rtp-time-diff

“max-rtcp-rtp-time-diff” gint

The maximum amount of time in ms that the RTP time in the RTCP SRs is allowed to be ahead of the last RTP packet we received. Use -1 to disable ignoring of RTCP packets.

Flags : Read / Write

Default value : 1000


max-ts-offset-adjustment

“max-ts-offset-adjustment” guint64

The maximum number of nanoseconds per frame that time offset may be adjusted with. This is used to avoid sudden large changes to time stamps.

Flags : Read / Write

Default value : 0


mode

“mode” Rtpjitter-buffer-mode

Control the buffering and timestamping mode used by the jitterbuffer.

Flags : Read / Write

Default value : slave (1)


name

“name” gchararray

The name of the object

Flags : Read / Write / Construct

Default value : NULL


parent

“parent” GstObject *

The parent of the object

Flags : Read / Write


percent

“percent” gint

The percent of the jitterbuffer that is filled.

Flags : Read

Default value : 0


rfc7273-sync

“rfc7273-sync” gboolean

Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided)

Flags : Read / Write

Default value : false


rtx-deadline

“rtx-deadline” gint

The deadline for a valid RTX request in ms.

How long the RTX RTCP will be valid for. When -1 is used, the size of the jitterbuffer will be used.

Flags : Read / Write

Default value : -1


rtx-delay

“rtx-delay” gint

When a packet did not arrive at the expected time, wait this extra amount of time before sending a retransmission event.

When -1 is used, the max jitter will be used as extra delay.

Flags : Read / Write

Default value : -1


rtx-delay-reorder

“rtx-delay-reorder” gint

Assume that a retransmission event should be sent when we see this much packet reordering.

When -1 is used, the value will be estimated based on observed packet reordering. When 0 is used packet reordering alone will not cause a retransmission event (Since 1.10).

Flags : Read / Write

Default value : 3


rtx-max-retries

“rtx-max-retries” gint

The maximum number of retries to request a retransmission.

This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested. When -1 is used, the number of retransmission request will not be limited.

Flags : Read / Write

Default value : -1


rtx-min-delay

“rtx-min-delay” guint

When a packet did not arrive at the expected time, wait at least this extra amount of time before sending a retransmission event.

Flags : Read / Write

Default value : 0


rtx-min-retry-timeout

“rtx-min-retry-timeout” gint

Minimum timeout between sending a transmission event in ms (-1 automatic)

Flags : Read / Write

Default value : -1


rtx-next-seqnum

“rtx-next-seqnum” gboolean

Estimate when the next packet should arrive and schedule a retransmission request for it. This is, when packet N arrives, a GstRTPRetransmission event is schedule for packet N+1. So it will be requested if it does not arrive at the expected time. The expected time is calculated using the dts of N and the packet spacing.

Flags : Read / Write

Default value : true


rtx-retry-period

“rtx-retry-period” gint

The amount of time to try to get a retransmission.

When -1 is used, the value will be estimated based on the jitterbuffer latency and the observed round trip time.

Flags : Read / Write

Default value : -1


rtx-retry-timeout

“rtx-retry-timeout” gint

Retry sending a transmission event after this timeout in ms (-1 automatic)

Flags : Read / Write

Default value : -1


rtx-stats-timeout

“rtx-stats-timeout” guint

The time to wait for a retransmitted packet after it has been considered lost in order to collect statistics (ms)

Flags : Read / Write

Default value : 1000


stats

“stats” GstStructure *

Various jitterbuffer statistics. This property returns a GstStructure with name application/x-rtp-jitterbuffer-stats with the following fields:

  • guint64 num-pushed: the number of packets pushed out.
  • guint64 num-lost: the number of packets considered lost.
  • guint64 num-late: the number of packets arriving too late.
  • guint64 num-duplicates: the number of duplicate packets.
  • guint64 rtx-count: the number of retransmissions requested.
  • guint64 rtx-success-count: the number of successful retransmissions.
  • gdouble rtx-per-packet: average number of RTX per packet.
  • guint64 rtx-rtt: average round trip time per RTX.

Flags : Read

Default value : application/x-rtp-jitterbuffer-stats, num-pushed=(guint64)0, num-lost=(guint64)0, num-late=(guint64)0, num-duplicates=(guint64)0, avg-jitter=(guint64)0, rtx-count=(guint64)0, rtx-success-count=(guint64)0, rtx-per-packet=(double)0, rtx-rtt=(guint64)0;


ts-offset

“ts-offset” gint64

Adjust GStreamer output buffer timestamps in the jitterbuffer with offset. This is mainly used to ensure interstream synchronisation.

Flags : Read / Write

Default value : 0


Named constants

Rtpjitter-buffer-mode

Members

none (0) – Only use RTP timestamps
slave (1) – Slave receiver to sender clock
buffer (2) – Do low/high watermark buffering
synced (4) – Synchronized sender and receiver clocks

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