# srtsrc

srtsrc is a network source that reads SRT packets from the network.

## Examples

     gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:7001" ! fakesink

This pipeline shows how to connect SRT server by setting GstSRTSrc:uri property.

     gst-launch-1.0 -v srtsrc uri="srt://:7001?mode=listener" ! fakesink

This pipeline shows how to wait SRT connection by setting GstSRTSrc:uri property.

     gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001?mode=rendez-vous" ! fakesink

This pipeline shows how to connect SRT server by setting GstSRTSrc:uri property and using the rendez-vous mode.

## Hierarchy

    GObject
        ╰──GInitiallyUnowned
            ╰──GstObject
                ╰──GstElement
                    ╰──GstBaseSrc
                        ╰──GstPushSrc
                            ╰──srtsrc

## Implemented interfaces

    GstURIHandler

## Factory details

Authors: – Justin Kim

Classification: – `Source/Network`

Rank – primary

Plugin – srt

Package – GStreamer Bad Plug-ins

## Pad Templates

### `src`

    ANY

Presence – always

Direction – src

Object type – GstPad

## Signals

### caller-added

    caller_added_callback (GstElement * gstsrtsink,
                           gint sock,
                           GSocketAddress * addr,
                           gpointer udata)

    def caller_added_callback (gstsrtsink, sock, addr, udata):
        #python callback for the 'caller-added' signal

    function caller_added_callback(gstsrtsink: GstElement * gstsrtsink, sock: gint sock, addr: GSocketAddress * addr, udata: gpointer udata): {
        // javascript callback for the 'caller-added' signal
    }

The given socket descriptor was added to srtsink.

Parameters:

`gstsrtsink` –

the srtsink element that emitted this signal

`sock` –

the client socket descriptor that was added to srtsink

`addr` –

the GSocketAddress that describes the sock

`udata` –
No description available

Flags: Run Last

### caller-removed

    caller_removed_callback (GstElement * gstsrtsink,
                             gint sock,
                             GSocketAddress * addr,
                             gpointer udata)

    def caller_removed_callback (gstsrtsink, sock, addr, udata):
        #python callback for the 'caller-removed' signal

    function caller_removed_callback(gstsrtsink: GstElement * gstsrtsink, sock: gint sock, addr: GSocketAddress * addr, udata: gpointer udata): {
        // javascript callback for the 'caller-removed' signal
    }

The given socket descriptor was removed from srtsink.

Parameters:

`gstsrtsink` –

the srtsink element that emitted this signal

`sock` –

the client socket descriptor that was added to srtsink

`addr` –

the GSocketAddress that describes the sock

`udata` –
No description available

Flags: Run Last

## Properties

### latency

    “latency” gint

The maximum accepted transmission latency.

Flags : Read / Write

Default value : 125

### localaddress

    “localaddress” gchararray

The address to bind when GstSRTSrc:mode is listener or rendezvous. This property can be set by URI parameters.

Flags : Read / Write

Default value : NULL

### localport

    “localport” guint

The local port to bind when GstSRTSrc:mode is listener or rendezvous. This property can be set by URI parameters.

Flags : Read / Write

Default value : 7001

### mode

    “mode” GstSRTConnectionMode *

The SRT connection mode. This property can be set by URI parameters.

Flags : Read / Write

Default value : caller (1)

### passphrase

    “passphrase” gchararray

The password for the encrypted transmission. This property can be set by URI parameters.

Flags : Read / Write

### pbkeylen

    “pbkeylen” GstSRTKeyLength *

The crypto key length. This property can be set by URI parameters.

Flags : Read / Write

Default value : no-key (0)

### poll-timeout

    “poll-timeout” gint

The polling timeout used when srt poll is started. Even if the default value indicates infinite waiting, it can be cancellable according to GstState This property can be set by URI parameters.

Flags : Read / Write

Default value : -1

### stats

    “stats” GstStructure *

The statistics from SRT.

Flags : Read

Default value :

    application/x-srt-statistics, bytes-received-total=(guint64)0;

### streamid

    “streamid” gchararray

The stream id for the SRT access control.

Flags : Read / Write

Default value : NULL

### uri

    “uri” gchararray

The URI used by SRT connection. User can specify SRT specific options by URI parameters. Refer to Mediun: SRT

Flags : Read / Write

Default value : srt://127.0.0.1:7001

### wait-for-connection

    “wait-for-connection” gboolean

Block the stream until a client connects

Flags : Read / Write

Default value : true

## Named constants

### GstSRTConnectionMode

#### Members

`none` (0) – GST_SRT_CONNECTION_MODE_NONE
`caller` (1) – GST_SRT_CONNECTION_MODE_CALLER
`listener` (2) – GST_SRT_CONNECTION_MODE_LISTENER
`rendezvous` (3) – GST_SRT_CONNECTION_MODE_RENDEZVOUS

### GstSRTKeyLength

#### Members

`no-key` (0) – GST_SRT_KEY_LENGTH_NO_KEY
`0` (0) – GST_SRT_KEY_LENGTH_0
`16` (16) – GST_SRT_KEY_LENGTH_16
`24` (24) – GST_SRT_KEY_LENGTH_24
`32` (32) – GST_SRT_KEY_LENGTH_32

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