## rtpjitterbuffer rtpjitterbuffer — A buffer that deals with network jitter and other transmission faults |

## Synopsis

    struct              GstRtpJitterBuffer;

## Object Hierarchy

      GObject
       +----GInitiallyUnowned
             +----GstObject
                   +----GstElement
                         +----GstRtpJitterBuffer

## Properties

      "do-lost"                  gboolean              : Read / Write
      "drop-on-latency"          gboolean              : Read / Write
      "latency"                  guint                 : Read / Write
      "ts-offset"                gint64                : Read / Write
      "mode"                     RTPJitterBufferMode   : Read / Write
      "percent"                  gint                  : Read
      "do-retransmission"        gboolean              : Read / Write
      "rtx-delay"                gint                  : Read / Write
      "rtx-delay-reorder"        gint                  : Read / Write
      "rtx-retry-period"         gint                  : Read / Write
      "rtx-retry-timeout"        gint                  : Read / Write

## Signals

      "clear-pt-map"                                   : Action
      "handle-sync"                                    : Run Last
      "on-npt-stop"                                    : Run Last
      "request-pt-map"                                 : Run Last
      "set-active"                                     : Action

## Description

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 incomming 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 abonormally 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

    1
    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.

Last reviewed on 2007-05-28 (0.10.5)

## Synopsis

### Element Information

plugin | rtpmanager
author | Philippe Kalaf <philippe.kalaf@collabora.co.uk>, Wim Taymans <wim.taymans@gmail.com>
class | Filter/Network/RTP

### Element Pads

name | sink
direction | sink
presence | always
details | application/x-rtp, clock-rate=(int)[ 1, 2147483647 ]

name | sink_rtcp
direction | sink
presence | request
details | application/x-rtcp

name | src
direction | source
presence | always
details | application/x-rtp

## Details

### struct GstRtpJitterBuffer

    struct GstRtpJitterBuffer;

Opaque jitterbuffer structure.

## Property Details

### The `"do-lost"` property

      "do-lost"                  gboolean              : Read / Write

Send an event downstream when a packet is lost.

Default value: FALSE

### The `"drop-on-latency"` property

      "drop-on-latency"          gboolean              : Read / Write

Tells the jitterbuffer to never exceed the given latency in size.

Default value: FALSE

### The `"latency"` property

      "latency"                  guint                 : Read / Write

Amount of ms to buffer.

Default value: 200

### The `"ts-offset"` property

      "ts-offset"                gint64                : Read / Write

Adjust buffer timestamps with offset in nanoseconds.

Default value: 0

### The `"mode"` property

      "mode"                     RTPJitterBufferMode   : Read / Write

Control the buffering algorithm in use.

Default value: Slave receiver to sender clock

### The `"percent"` property

      "percent"                  gint                  : Read

The buffer filled percent.

Allowed values: [0,100]

Default value: 0

### The `"do-retransmission"` property

      "do-retransmission"        gboolean              : Read / Write

Send retransmission events upstream when a packet is late.

Default value: FALSE

### The `"rtx-delay"` property

      "rtx-delay"                gint                  : Read / Write

Extra time in ms to wait before sending retransmission event (-1 automatic).

Allowed values: >= G_MAXULONG

Default value: 20

### The `"rtx-delay-reorder"` property

      "rtx-delay-reorder"        gint                  : Read / Write

Sending retransmission event when this much reordering (-1 automatic).

Allowed values: >= G_MAXULONG

Default value: 3

### The `"rtx-retry-period"` property

      "rtx-retry-period"         gint                  : Read / Write

Try to get a retransmission for this many ms (-1 automatic).

Allowed values: >= G_MAXULONG

Default value: 160

### The `"rtx-retry-timeout"` property

      "rtx-retry-timeout"        gint                  : Read / Write

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

Allowed values: >= G_MAXULONG

Default value: 40

## Signal Details

### The `"clear-pt-map"` signal

    void                user_function                      (GstRtpJitterBuffer *buffer,
                                                            gpointer            user_data)      : Action

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

`buffer` : | the object which received the signal
`user_data` : | user data set when the signal handler was connected.

### The `"handle-sync"` signal

    void                user_function                      (GstRtpJitterBuffer *buffer,
                                                            GstStructure       *struct,
                                                            gpointer            user_data)      : Run Last

Be notified of new sync values.

`buffer` : | the object which received the signal
`struct` : | a GstStructure containing sync values.
`user_data` : | user data set when the signal handler was connected.

### The `"on-npt-stop"` signal

    void                user_function                      (GstRtpJitterBuffer *buffer,
                                                            gpointer            user_data)      : Run Last

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

`buffer` : | the object which received the signal
`user_data` : | user data set when the signal handler was connected.

### The `"request-pt-map"` signal

    GstCaps*            user_function                      (GstRtpJitterBuffer *buffer,
                                                            guint               pt,
                                                            gpointer            user_data)      : Run Last

Request the payload type as GstCaps for `pt`.

`buffer` : | the object which received the signal
`pt` : | the pt
`user_data` : | user data set when the signal handler was connected.

### The `"set-active"` signal

    guint64             user_function                      (GstRtpJitterBuffer *buffer,
                                                            gboolean            Returns,
                                                            guint64             arg2,
                                                            gpointer            user_data)      : Action

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

`buffer` : | the object which received the signal
`user_data` : | user data set when the signal handler was connected.
Returns : | the time of the last pushed packet.

Since 0.10.19