GstRtspServer.RTSPMedia

g GObject.Object GObject.Object GstRtspServer.RTSPMedia GstRtspServer.RTSPMedia GObject.Object->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:
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:
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:
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:
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:
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:
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:
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:
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:
GstRtspServer.RTSPMedia.signals.new_stream(r_t_s_p_media, object)
Signal Name:

new-stream

Flags:

RUN_LAST

Parameters:
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:
GstRtspServer.RTSPMedia.signals.target_state(r_t_s_p_media, object)
Signal Name:

target-state

Flags:

RUN_LAST

Parameters:
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