GstRtspServer.RTSPMedia¶

Subclasses:	`GstRtspServer.RTSPOnvifMedia`

Methods¶

Inherited:	GObject.Object (37)
Structs:	GObject.ObjectClass (5)

class	`new` (element)
	`collect_streams` ()
	`complete_pipeline` (transports)
	`create_stream` (payloader, pad)
	`find_stream` (control)
	`get_address_pool` ()
	`get_base_time` ()
	`get_buffer_size` ()
	`get_clock` ()
	`get_do_retransmission` ()
	`get_dscp_qos` ()
	`get_element` ()
	`get_latency` ()
	`get_max_mcast_ttl` ()
	`get_multicast_iface` ()
	`get_permissions` ()
	`get_profiles` ()
	`get_protocols` ()
	`get_publish_clock_mode` ()
	`get_range_string` (play, unit)
	`get_rate_control` ()
	`get_rates` ()
	`get_retransmission_time` ()
	`get_status` ()
	`get_stream` (idx)
	`get_suspend_mode` ()
	`get_time_provider` (address, port)
	`get_transport_mode` ()
	`handle_sdp` (sdp)
	`has_completed_sender` ()
	`is_bind_mcast_address` ()
	`is_eos_shutdown` ()
	`is_receive_only` ()
	`is_reusable` ()
	`is_shared` ()
	`is_stop_on_disconnect` ()
	`is_time_provider` ()
	`lock` ()
	`n_streams` ()
	`prepare` (thread)
	`seek` (range)
	`seek_full` (range, flags)
	`seek_trickmode` (range, flags, rate, trickmode_interval)
	`seekable` ()
	`set_address_pool` (pool)
	`set_bind_mcast_address` (bind_mcast_addr)
	`set_buffer_size` (size)
	`set_clock` (clock)
	`set_do_retransmission` (do_retransmission)
	`set_dscp_qos` (dscp_qos)
	`set_eos_shutdown` (eos_shutdown)
	`set_latency` (latency)
	`set_max_mcast_ttl` (ttl)
	`set_multicast_iface` (multicast_iface)
	`set_permissions` (permissions)
	`set_pipeline_state` (state)
	`set_profiles` (profiles)
	`set_protocols` (protocols)
	`set_publish_clock_mode` (mode)
	`set_rate_control` (enabled)
	`set_retransmission_time` (time)
	`set_reusable` (reusable)
	`set_shared` (shared)
	`set_state` (state, transports)
	`set_stop_on_disconnect` (stop_on_disconnect)
	`set_suspend_mode` (mode)
	`set_transport_mode` (mode)
	`setup_sdp` (sdp, info)
	`suspend` ()
	`take_pipeline` (pipeline)
	`unlock` ()
	`unprepare` ()
	`unsuspend` ()
	`use_time_provider` (time_provider)

Virtual Methods¶

Inherited:	GObject.Object (7)

	`do_convert_range` (range, unit)
	`do_handle_message` (message)
	`do_handle_sdp` (sdp)
	`do_new_state` (state)
	`do_new_stream` (stream)
	`do_prepare` (thread)
	`do_prepared` ()
	`do_query_position` (position)
	`do_query_stop` (stop)
	`do_removed_stream` (stream)
	`do_setup_rtpbin` (rtpbin)
	`do_setup_sdp` (sdp, info)
	`do_suspend` ()
	`do_target_state` (state)
	`do_unprepare` ()
	`do_unprepared` ()
	`do_unsuspend` ()

Properties¶

Name	Type	Flags	Short Description
`bind-mcast-address`	`bool`	r/w	Whether the multicast sockets should be bound to multicast addresses or INADDR_ANY
`buffer-size`	`int`	r/w	The kernel UDP buffer size to use
`clock`	`Gst.Clock`	r/w	Clock to be used by the media pipeline
`dscp-qos`	`int`	r/w	The IP DSCP field to use for each related stream
`element`	`Gst.Element`	r/w/co	The `Gst.Bin` to use for streaming the media
`eos-shutdown`	`bool`	r/w	Send an EOS event to the pipeline before unpreparing
`latency`	`int`	r/w	Latency used for receiving media in milliseconds
`max-mcast-ttl`	`int`	r/w	The maximum time-to-live value of outgoing multicast packets
`profiles`	`GstRtsp.RTSPProfile`	r/w	Allowed transfer profiles
`protocols`	`GstRtsp.RTSPLowerTrans`	r/w	Allowed lower transport protocols
`reusable`	`bool`	r/w	If this media pipeline can be reused after an unprepare
`shared`	`bool`	r/w	If this media pipeline can be shared
`stop-on-disconnect`	`bool`	r/w	If this media pipeline should be stopped when a client disconnects without TEARDOWN
`suspend-mode`	`GstRtspServer.RTSPSuspendMode`	r/w	How to suspend the media in PAUSED
`time-provider`	`bool`	r/w	Use a NetTimeProvider for clients
`transport-mode`	`GstRtspServer.RTSPTransportMode`	r/w	If this media pipeline can be used for PLAY or RECORD

Signals¶

Inherited:	GObject.Object (1)

Name	Short Description
`new-state`
`new-stream`
`prepared`
`removed-stream`
`target-state`
`unprepared`

Fields¶

Inherited:	GObject.Object (1)

Name	Type	Access	Description
parent	`GObject.Object`	r

Class Details¶

class GstRtspServer.RTSPMedia(**kwargs)¶

Bases:	`GObject.Object`
Abstract:	No
Structure:	`GstRtspServer.RTSPMediaClass`

A class that contains the GStreamer element along with a list of GstRtspServer.RTSPStream objects that can produce GstRtsp.data.

This object is usually created from a GstRtspServer.RTSPMediaFactory.

classmethod new(element)[source]¶

Parameters:	element (`Gst.Element`) – a `Gst.Element`
Returns:	a new `GstRtspServer.RTSPMedia` object.
Return type:	`GstRtspServer.RTSPMedia`

Create a new GstRtspServer.RTSPMedia instance. element is the bin element that provides the different streams. The GstRtspServer.RTSPMedia object contains the element to produce RTP GstRtsp.data for one or more related (audio/video/..) streams.

Ownership is taken of element.

collect_streams()[source]¶

Find all payloader elements, they should be named pay\%d in the element of self, and create GstRtspServer.RTSPStreams for them.

Collect all dynamic elements, named dynpay\%d, and add them to the list of dynamic elements.

Find all depayloader elements, they should be named depay\%d in the element of self, and create GstRtspServer.RTSPStreams for them.

complete_pipeline(transports)[source]¶

Parameters:	transports ([`GstRtsp.RTSPTransport`]) – a list of `GstRtsp.RTSPTransport`
Returns:	`True` if the media pipeline has been sucessfully updated.
Return type:	`bool`

Add a receiver and sender parts to the pipeline based on the transport from SETUP.

New in version 1.14.

create_stream(payloader, pad)[source]¶

Parameters:	payloader (`Gst.Element`) – a `Gst.Element` pad (`Gst.Pad`) – a `Gst.Pad`
Returns:	a new `GstRtspServer.RTSPStream` that remains valid for as long as self exists.
Return type:	`GstRtspServer.RTSPStream`

Create a new stream in self that provides RTP GstRtsp.data on pad. pad should be a pad of an element inside self->element.

find_stream(control)[source]¶

Parameters:	control (`str`) – the control of the stream
Returns:	the `GstRtspServer.RTSPStream` with control uri control or `None` when a stream with that control did not exist.
Return type:	`GstRtspServer.RTSPStream` or `None`

Find a stream in self with control as the control uri.

get_address_pool()[source]¶

Returns:	the `GstRtspServer.RTSPAddressPool` of self. `GObject.Object.unref`() after usage.
Return type:	`GstRtspServer.RTSPAddressPool` or `None`

Get the GstRtspServer.RTSPAddressPool used as the address pool of self.

get_base_time()[source]¶

Returns:	the base_time used by self.
Return type:	`int`

Get the base_time that is used by the pipeline in self.

self must be prepared before this method returns a valid base_time.

get_buffer_size()[source]¶

Returns:	the kernel UDP buffer size.
Return type:	`int`

Get the kernel UDP buffer size.

get_clock()[source]¶

Returns:	the `Gst.Clock` used by self. unref after usage.
Return type:	`Gst.Clock` or `None`

Get the clock that is used by the pipeline in self.

self must be prepared before this method returns a valid clock object.

get_do_retransmission()[source]¶

Returns:	Whether retransmission requests will be sent
Return type:	`bool`

New in version 1.16.

get_dscp_qos()[source]¶

Returns:	the DSCP QoS value of attached streams or -1 if disabled.
Return type:	`int`

Get the configured DSCP QoS of attached media.

New in version 1.18.

get_element()[source]¶

Returns:	a `Gst.Element`. Unref after usage.
Return type:	`Gst.Element`

Get the element that was used when constructing self.

get_latency()[source]¶

Returns:	latency in milliseconds
Return type:	`int`

Get the latency that is used for receiving media.

get_max_mcast_ttl()[source]¶

Returns:	the maximum time-to-live value of outgoing multicast packets.
Return type:	`int`

Get the the maximum time-to-live value of outgoing multicast packets.

New in version 1.16.

get_multicast_iface()[source]¶

Returns:	the multicast interface for self. `GLib.free`() after usage.
Return type:	`str` or `None`

Get the multicast interface used for self.

get_permissions()[source]¶

Returns:	a `GstRtspServer.RTSPPermissions` object, unref after usage.
Return type:	`GstRtspServer.RTSPPermissions` or `None`

Get the permissions object from self.

get_profiles()[source]¶

Returns:	a `GstRtsp.RTSPProfile`
Return type:	`GstRtsp.RTSPProfile`

Get the allowed profiles of self.

get_protocols()[source]¶

Returns:	a `GstRtsp.RTSPLowerTrans`
Return type:	`GstRtsp.RTSPLowerTrans`

Get the allowed protocols of self.

get_publish_clock_mode()[source]¶

Returns:	The `GstRtspServer.RTSPPublishClockMode`
Return type:	`GstRtspServer.RTSPPublishClockMode`

Gets if and how the media clock should be published according to RFC7273.

New in version 1.8.

get_range_string(play, unit)[source]¶

Parameters:	play (`bool`) – for the PLAY `GstRtsp.request` unit (`GstRtsp.RTSPRangeUnit`) – the unit to use for the string
Returns:	The range as a string, `GLib.free`() after usage.
Return type:	`str` or `None`

Get the current range as a string. self must be prepared with GstRtspServer.RTSPMedia.prepare ().

get_rate_control()[source]¶

Returns:	whether self will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec.
Return type:	`bool`

New in version 1.18.

get_rates()[source]¶

Returns:

False if looking up the rate and applied rate failed. Otherwise True is returned and rate and applied_rate are set to the rate and applied_rate of the current segment.

rate:	the rate of the current segment
applied_rate:	the applied_rate of the current segment

Return type: (bool, rate: float, applied_rate: float)

Get the rate and applied_rate of the current segment.

New in version 1.18.

get_retransmission_time()[source]¶

Returns:	the amount of time to store retransmission `GstRtsp.data`.
Return type:	`int`

Get the amount of time to store retransmission GstRtsp.data.

get_status()[source]¶

Returns:	the status of self.
Return type:	`GstRtspServer.RTSPMediaStatus`

Get the status of self. When self is busy preparing, this function waits until self is prepared or in error.

get_stream(idx)[source]¶

Parameters:	idx (`int`) – the stream index
Returns:	the `GstRtspServer.RTSPStream` at index idx or `None` when a stream with that index did not exist.
Return type:	`GstRtspServer.RTSPStream` or `None`

Retrieve the stream with index idx from self.

get_suspend_mode()[source]¶

Returns:	`GstRtspServer.RTSPSuspendMode`.
Return type:	`GstRtspServer.RTSPSuspendMode`

Get how self will be suspended.

get_time_provider(address, port)[source]¶

Parameters:	address (`str` or `None`) – an address or `None` port (`int`) – a port or 0
Returns:	the `GstNet.NetTimeProvider` of self.
Return type:	`GstNet.NetTimeProvider`

Get the GstNet.NetTimeProvider for the clock used by self. The time provider will listen on address and port for client time requests.

get_transport_mode()[source]¶

Returns:	The transport mode.
Return type:	`GstRtspServer.RTSPTransportMode`

Check if the pipeline for self can be used for PLAY or RECORD methods.

handle_sdp(sdp)[source]¶

Parameters:	sdp (`GstSdp.SDPMessage`) – a `GstSdp.SDPMessage`
Returns:	`True` on success.
Return type:	`bool`

Configure an SDP on self for receiving streams

has_completed_sender()[source]¶

Returns:	whether self has at least one complete sender stream.
Return type:	`bool`

See GstRtspServer.RTSPStream.is_complete(), GstRtspServer.RTSPStream.is_sender().

New in version 1.18.

is_bind_mcast_address()[source]¶

Returns:	`True` if multicast sockets are configured to be bound to multicast addresses.
Return type:	`bool`

Check if multicast sockets are configured to be bound to multicast addresses.

New in version 1.16.

is_eos_shutdown()[source]¶

Returns:	`True` if the media will send EOS before unpreparing.
Return type:	`bool`

Check if the pipeline for self will send an EOS down the pipeline before unpreparing.

is_receive_only()[source]¶

Returns:	`True` if self is receive-only, `False` otherwise.
Return type:	`bool`

New in version 1.18.

is_reusable()[source]¶

Returns:	`True` if the media can be reused
Return type:	`bool`

Check if the pipeline for self can be reused after an unprepare.

is_shared()[source]¶

Returns:	`True` if the media can be shared between clients.
Return type:	`bool`

Check if the pipeline for self can be shared between multiple clients.

is_stop_on_disconnect()[source]¶

Returns:	`True` if the media will be stopped when a client disconnects without sending TEARDOWN.
Return type:	`bool`

Check if the pipeline for self will be stopped when a client disconnects without sending TEARDOWN.

is_time_provider()[source]¶

Returns:	`True` if self can provide a `GstNet.NetTimeProvider`.
Return type:	`bool`

Check if self can provide a GstNet.NetTimeProvider for its pipeline clock.

Use GstRtspServer.RTSPMedia.get_time_provider() to get the network clock.

lock()[source]¶

Lock the entire media. This is needed by callers such as rtsp_client to protect the media when it is shared by many clients. The lock prevents that concurrent clients alters the shared media, while one client already is working with it. Typically the lock is taken in external RTSP API calls that uses shared media such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.

As best practice take the lock as soon as the function get hold of a shared media object. Release the lock right before the function returns.

New in version 1.18.

n_streams()[source]¶

Returns:	The number of streams.
Return type:	`int`

Get the number of streams in this media.

prepare(thread)[source]¶

Parameters:	thread (`GstRtspServer.RTSPThread` or `None`) – a `GstRtspServer.RTSPThread` to run the bus handler or `None`
Returns:	`True` on success.
Return type:	`bool`

Prepare self for streaming. This function will create the objects to manage the streaming. A pipeline must have been set on self with GstRtspServer.RTSPMedia.take_pipeline().

It will preroll the pipeline and collect vital information about the streams such as the duration.

seek(range)[source]¶

Parameters:	range (`GstRtsp.RTSPTimeRange`) – a `GstRtsp.RTSPTimeRange`
Returns:	`True` on success.
Return type:	`bool`

Seek the pipeline of self to range. self must be prepared with GstRtspServer.RTSPMedia.prepare().

seek_full(range, flags)[source]¶

Parameters:	range (`GstRtsp.RTSPTimeRange`) – a `GstRtsp.RTSPTimeRange` flags (`Gst.SeekFlags`) – The minimal set of `Gst.SeekFlags` to use
Returns:	`True` on success.
Return type:	`bool`

Seek the pipeline of self to range with the given flags. self must be prepared with GstRtspServer.RTSPMedia.prepare().

New in version 1.18.

seek_trickmode(range, flags, rate, trickmode_interval)[source]¶

Parameters:	range (`GstRtsp.RTSPTimeRange`) – a `GstRtsp.RTSPTimeRange` flags (`Gst.SeekFlags`) – The minimal set of `Gst.SeekFlags` to use rate (`float`) – the rate to use in the seek trickmode_interval (`int`) – The trickmode interval to use for KEY_UNITS trick mode
Returns:	`True` on success.
Return type:	`bool`

Seek the pipeline of self to range with the given flags and rate, and trickmode_interval. self must be prepared with GstRtspServer.RTSPMedia.prepare(). In order to perform the seek operation, the pipeline must contain all needed transport parts (transport sinks).

New in version 1.18.

seekable()[source]¶

Returns:	-1 if the stream is not seekable, 0 if seekable only to the beginning and > 0 to indicate the longest duration between any two random access points. `GObject.G_MAXINT64` means any value is possible.
Return type:	`int`

Check if the pipeline for self seek and up to what point in time, it can seek.

New in version 1.14.

set_address_pool(pool)[source]¶

Parameters:	pool (`GstRtspServer.RTSPAddressPool` or `None`) – a `GstRtspServer.RTSPAddressPool`

configure pool to be used as the address pool of self.

set_bind_mcast_address(bind_mcast_addr)[source]¶

Parameters:	bind_mcast_addr (`bool`) – the new value

Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY.

New in version 1.16.

set_buffer_size(size)[source]¶

Parameters:	size (`int`) – the new value

Set the kernel UDP buffer size.

set_clock(clock)[source]¶

Parameters:	clock (`Gst.Clock` or `None`) – `Gst.Clock` to be used

Configure the clock used for the media.

set_do_retransmission(do_retransmission)[source]¶

Parameters:	do_retransmission (`bool`) –

Set whether retransmission requests will be sent

New in version 1.16.

set_dscp_qos(dscp_qos)[source]¶

Parameters:	dscp_qos (`int`) – a new dscp qos value (0-63, or -1 to disable)

Configure the dscp qos of attached streams to dscp_qos.

New in version 1.18.

set_eos_shutdown(eos_shutdown)[source]¶

Parameters:	eos_shutdown (`bool`) – the new value

Set or unset if an EOS event will be sent to the pipeline for self before it is unprepared.

set_latency(latency)[source]¶

Parameters:	latency (`int`) – latency in milliseconds

Configure the latency used for receiving media.

set_max_mcast_ttl(ttl)[source]¶

Parameters:	ttl (`int`) – the new multicast ttl value
Returns:	`True` if the requested ttl has been set successfully.
Return type:	`bool`

Set the maximum time-to-live value of outgoing multicast packets.

New in version 1.16.

set_multicast_iface(multicast_iface)[source]¶

Parameters:	multicast_iface (`str` or `None`) – a multicast interface name

configure multicast_iface to be used for self.

set_permissions(permissions)[source]¶

Parameters:	permissions (`GstRtspServer.RTSPPermissions` or `None`) – a `GstRtspServer.RTSPPermissions`

Set permissions on self.

set_pipeline_state(state)[source]¶

Parameters:	state (`Gst.State`) – the target state of the pipeline

Set the state of the pipeline managed by self to state

set_profiles(profiles)[source]¶

Parameters:	profiles (`GstRtsp.RTSPProfile`) – the new flags

Configure the allowed lower transport for self.

set_protocols(protocols)[source]¶

Parameters:	protocols (`GstRtsp.RTSPLowerTrans`) – the new flags

Configure the allowed lower transport for self.

set_publish_clock_mode(mode)[source]¶

Parameters:	mode (`GstRtspServer.RTSPPublishClockMode`) – the clock publish mode

Sets if and how the media clock should be published according to RFC7273.

New in version 1.8.

set_rate_control(enabled)[source]¶

Parameters:	enabled (`bool`) –

Define whether self will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec.

New in version 1.18.

set_retransmission_time(time)[source]¶

Parameters:	time (`int`) – the new value

Set the amount of time to store retransmission packets.

set_reusable(reusable)[source]¶

Parameters:	reusable (`bool`) – the new value

Set or unset if the pipeline for self can be reused after the pipeline has been unprepared.

set_shared(shared)[source]¶

Parameters:	shared (`bool`) – the new value

Set or unset if the pipeline for self can be shared will multiple clients. When shared is True, client requests for this media will share the media pipeline.

set_state(state, transports)[source]¶

Parameters:	state (`Gst.State`) – the target state of the media transports ([`GstRtspServer.RTSPStreamTransport`]) – a `GLib.PtrArray` of `GstRtspServer.RTSPStreamTransport` pointers
Returns:	`True` on success.
Return type:	`bool`

Set the state of self to state and for the transports in transports.

self must be prepared with GstRtspServer.RTSPMedia.prepare();

set_stop_on_disconnect(stop_on_disconnect)[source]¶

Parameters:	stop_on_disconnect (`bool`) – the new value

Set or unset if the pipeline for self should be stopped when a client disconnects without sending TEARDOWN.

set_suspend_mode(mode)[source]¶

Parameters:	mode (`GstRtspServer.RTSPSuspendMode`) – the new `GstRtspServer.RTSPSuspendMode`

Control how @ media will be suspended after the SDP has been generated and after a PAUSE GstRtsp.request has been performed.

Media must be unprepared when setting the suspend mode.

set_transport_mode(mode)[source]¶

Parameters:	mode (`GstRtspServer.RTSPTransportMode`) – the new value

Sets if the media pipeline can work in PLAY or RECORD mode

setup_sdp(sdp, info)[source]¶

Parameters:	sdp (`GstSdp.SDPMessage`) – a `GstSdp.SDPMessage` info (`GstRtspServer.SDPInfo`) – a `GstRtspServer.SDPInfo`
Returns:	`True` on success.
Return type:	`bool`

Add self specific info to sdp. info is used to configure the connection information in the SDP.

suspend()[source]¶

Returns:	`True` on success.
Return type:	`bool`

Suspend self. The state of the pipeline managed by self is set to Gst.State.NULL but all streams are kept. self can be prepared again with GstRtspServer.RTSPMedia.unsuspend()

self must be prepared with GstRtspServer.RTSPMedia.prepare();

take_pipeline(pipeline)[source]¶

Parameters:	pipeline (`Gst.Pipeline`) – a `Gst.Pipeline`

Set pipeline as the Gst.Pipeline for self. Ownership is taken of pipeline.

unlock()[source]¶: Unlock the media.

New in version 1.18.

unprepare()[source]¶

Returns:	`True` on success.
Return type:	`bool`

Unprepare self. After this call, the media should be prepared again before it can be used again. If the media is set to be non-reusable, a new instance must be created.

unsuspend()[source]¶

Returns:	`True` on success.
Return type:	`bool`

Unsuspend self if it was in a suspended state. This method does nothing when the media was not in the suspended state.

use_time_provider(time_provider)[source]¶

Parameters:	time_provider (`bool`) – if a `GstNet.NetTimeProvider` should be used

Set self to provide a GstNet.NetTimeProvider.

do_convert_range(range, unit) virtual¶

Parameters:	range (`GstRtsp.RTSPTimeRange`) – unit (`GstRtsp.RTSPRangeUnit`) –
Return type:	`bool`

do_handle_message(message) virtual¶

Parameters:	message (`Gst.Message`) –
Return type:	`bool`

do_handle_sdp(sdp) virtual¶

Parameters:	sdp (`GstSdp.SDPMessage`) – a `GstSdp.SDPMessage`
Returns:	`True` on success.
Return type:	`bool`

Configure an SDP on media for receiving streams

do_new_state(state) virtual¶

Parameters:	state (`Gst.State`) –

do_new_stream(stream) virtual¶

Parameters:	stream (`GstRtspServer.RTSPStream`) –

do_prepare(thread) virtual¶

Parameters:	thread (`GstRtspServer.RTSPThread` or `None`) – a `GstRtspServer.RTSPThread` to run the bus handler or `None`
Returns:	`True` on success.
Return type:	`bool`

Prepare media for streaming. This function will create the objects to manage the streaming. A pipeline must have been set on media with GstRtspServer.RTSPMedia.take_pipeline().

It will preroll the pipeline and collect vital information about the streams such as the duration.

do_prepared() virtual¶

do_query_position(position) virtual¶

Parameters:	position (`int`) –
Return type:	`bool`

do_query_stop(stop) virtual¶

Parameters:	stop (`int`) –
Return type:	`bool`

do_removed_stream(stream) virtual¶

Parameters:	stream (`GstRtspServer.RTSPStream`) –

do_setup_rtpbin(rtpbin) virtual¶

Parameters:	rtpbin (`Gst.Element`) –
Return type:	`bool`

do_setup_sdp(sdp, info) virtual¶

Parameters:	sdp (`GstSdp.SDPMessage`) – a `GstSdp.SDPMessage` info (`GstRtspServer.SDPInfo`) – a `GstRtspServer.SDPInfo`
Returns:	`True` on success.
Return type:	`bool`

Add media specific info to sdp. info is used to configure the connection information in the SDP.

do_suspend() virtual¶

Returns:	`True` on success.
Return type:	`bool`

Suspend media. The state of the pipeline managed by media is set to Gst.State.NULL but all streams are kept. media can be prepared again with GstRtspServer.RTSPMedia.unsuspend()

media must be prepared with GstRtspServer.RTSPMedia.prepare();

do_target_state(state) virtual¶

Parameters:	state (`Gst.State`) –

do_unprepare() virtual¶

Returns:	`True` on success.
Return type:	`bool`

Unprepare media. After this call, the media should be prepared again before it can be used again. If the media is set to be non-reusable, a new instance must be created.

do_unprepared() virtual¶

do_unsuspend() virtual¶

Returns:	`True` on success.
Return type:	`bool`

Unsuspend media if it was in a suspended state. This method does nothing when the media was not in the suspended state.

Signal Details¶

GstRtspServer.RTSPMedia.signals.new_state(r_t_s_p_media, object)¶

Signal Name:	`new-state`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_media (`GstRtspServer.RTSPMedia`) – The object which received the signal object (`int`) –

GstRtspServer.RTSPMedia.signals.new_stream(r_t_s_p_media, object)¶

Signal Name:	`new-stream`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_media (`GstRtspServer.RTSPMedia`) – The object which received the signal object (`GstRtspServer.RTSPStream`) –

GstRtspServer.RTSPMedia.signals.prepared(r_t_s_p_media)¶

Signal Name:	`prepared`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_media (`GstRtspServer.RTSPMedia`) – The object which received the signal

GstRtspServer.RTSPMedia.signals.removed_stream(r_t_s_p_media, object)¶

Signal Name:	`removed-stream`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_media (`GstRtspServer.RTSPMedia`) – The object which received the signal object (`GstRtspServer.RTSPStream`) –

GstRtspServer.RTSPMedia.signals.target_state(r_t_s_p_media, object)¶

Signal Name:	`target-state`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_media (`GstRtspServer.RTSPMedia`) – The object which received the signal object (`int`) –

GstRtspServer.RTSPMedia.signals.unprepared(r_t_s_p_media)¶

Signal Name:	`unprepared`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_media (`GstRtspServer.RTSPMedia`) – The object which received the signal

Property Details¶

GstRtspServer.RTSPMedia.props.bind_mcast_address¶

Name:	`bind-mcast-address`
Type:	`bool`
Default Value:	`False`
Flags:	`READABLE`, `WRITABLE`

Whether the multicast sockets should be bound to multicast addresses or INADDR_ANY

GstRtspServer.RTSPMedia.props.buffer_size¶

Name:	`buffer-size`
Type:	`int`
Default Value:	`524288`
Flags:	`READABLE`, `WRITABLE`

The kernel UDP buffer size to use

GstRtspServer.RTSPMedia.props.clock¶

Name:	`clock`
Type:	`Gst.Clock`
Default Value:	`None`
Flags:	`READABLE`, `WRITABLE`

Clock to be used by the media pipeline

GstRtspServer.RTSPMedia.props.dscp_qos¶

Name:	`dscp-qos`
Type:	`int`
Default Value:	`-1`
Flags:	`READABLE`, `WRITABLE`

The IP DSCP field to use for each related stream

GstRtspServer.RTSPMedia.props.element¶

Name:	`element`
Type:	`Gst.Element`
Default Value:	`None`
Flags:	`READABLE`, `WRITABLE`, `CONSTRUCT_ONLY`

The Gst.Bin to use for streaming the media

GstRtspServer.RTSPMedia.props.eos_shutdown¶

Name:	`eos-shutdown`
Type:	`bool`
Default Value:	`False`
Flags:	`READABLE`, `WRITABLE`

Send an EOS event to the pipeline before unpreparing

GstRtspServer.RTSPMedia.props.latency¶

Name:	`latency`
Type:	`int`
Default Value:	`200`
Flags:	`READABLE`, `WRITABLE`

Latency used for receiving media in milliseconds

GstRtspServer.RTSPMedia.props.max_mcast_ttl¶

Name:	`max-mcast-ttl`
Type:	`int`
Default Value:	`255`
Flags:	`READABLE`, `WRITABLE`

The maximum time-to-live value of outgoing multicast packets

GstRtspServer.RTSPMedia.props.profiles¶

Name:	`profiles`
Type:	`GstRtsp.RTSPProfile`
Default Value:	`GstRtsp.RTSPProfile.AVP`
Flags:	`READABLE`, `WRITABLE`

Allowed transfer profiles

GstRtspServer.RTSPMedia.props.protocols¶

Name:	`protocols`
Type:	`GstRtsp.RTSPLowerTrans`
Default Value:	`GstRtsp.RTSPLowerTrans.UDP` \| `GstRtsp.RTSPLowerTrans.UDP_MCAST` \| `GstRtsp.RTSPLowerTrans.TCP`
Flags:	`READABLE`, `WRITABLE`

Allowed lower transport protocols

GstRtspServer.RTSPMedia.props.reusable¶

Name:	`reusable`
Type:	`bool`
Default Value:	`False`
Flags:	`READABLE`, `WRITABLE`

If this media pipeline can be reused after an unprepare

GstRtspServer.RTSPMedia.props.shared¶

Name:	`shared`
Type:	`bool`
Default Value:	`False`
Flags:	`READABLE`, `WRITABLE`

If this media pipeline can be shared

GstRtspServer.RTSPMedia.props.stop_on_disconnect¶

Name:	`stop-on-disconnect`
Type:	`bool`
Default Value:	`True`
Flags:	`READABLE`, `WRITABLE`

If this media pipeline should be stopped when a client disconnects without TEARDOWN

GstRtspServer.RTSPMedia.props.suspend_mode¶

Name:	`suspend-mode`
Type:	`GstRtspServer.RTSPSuspendMode`
Default Value:	`GstRtspServer.RTSPSuspendMode.NONE`
Flags:	`READABLE`, `WRITABLE`

How to suspend the media in PAUSED

GstRtspServer.RTSPMedia.props.time_provider¶

Name:	`time-provider`
Type:	`bool`
Default Value:	`False`
Flags:	`READABLE`, `WRITABLE`

Use a NetTimeProvider for clients

GstRtspServer.RTSPMedia.props.transport_mode¶

Name:	`transport-mode`
Type:	`GstRtspServer.RTSPTransportMode`
Default Value:	`GstRtspServer.RTSPTransportMode.PLAY`
Flags:	`READABLE`, `WRITABLE`

If this media pipeline can be used for PLAY or RECORD