GstRtspServer.RTSPStream¶

Subclasses:	None

Methods¶

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

class	`new` (idx, payloader, pad)
	`add_multicast_client_address` (destination, rtp_port, rtcp_port, family)
	`add_transport` (trans)
	`allocate_udp_sockets` (family, transport, use_client_settings)
	`complete_stream` (transport)
	`get_address_pool` ()
	`get_buffer_size` ()
	`get_caps` ()
	`get_control` ()
	`get_current_seqnum` ()
	`get_dscp_qos` ()
	`get_index` ()
	`get_joined_bin` ()
	`get_max_mcast_ttl` ()
	`get_mtu` ()
	`get_multicast_address` (family)
	`get_multicast_client_addresses` ()
	`get_multicast_iface` ()
	`get_profiles` ()
	`get_protocols` ()
	`get_pt` ()
	`get_publish_clock_mode` ()
	`get_rate_control` ()
	`get_rates` ()
	`get_retransmission_pt` ()
	`get_retransmission_time` ()
	`get_rtcp_multicast_socket` (family)
	`get_rtcp_socket` (family)
	`get_rtp_multicast_socket` (family)
	`get_rtp_socket` (family)
	`get_rtpinfo` ()
	`get_rtpsession` ()
	`get_server_port` (family)
	`get_sinkpad` ()
	`get_srcpad` ()
	`get_srtp_encoder` ()
	`get_ssrc` ()
	`get_ulpfec_enabled` ()
	`get_ulpfec_percentage` ()
	`get_ulpfec_pt` ()
	`handle_keymgmt` (keymgmt)
	`has_control` (control)
	`is_bind_mcast_address` ()
	`is_blocking` ()
	`is_client_side` ()
	`is_complete` ()
	`is_receiver` ()
	`is_sender` ()
	`is_transport_supported` (transport)
	`join_bin` (bin, rtpbin, state)
	`leave_bin` (bin, rtpbin)
	`query_position` ()
	`query_stop` ()
	`recv_rtcp` (buffer)
	`recv_rtp` (buffer)
	`remove_transport` (trans)
	`request_aux_receiver` (sessid)
	`request_aux_sender` (sessid)
	`request_ulpfec_decoder` (rtpbin, sessid)
	`request_ulpfec_encoder` (sessid)
	`reserve_address` (address, port, n_ports, ttl)
	`seekable` ()
	`set_address_pool` (pool)
	`set_bind_mcast_address` (bind_mcast_addr)
	`set_blocked` (blocked)
	`set_buffer_size` (size)
	`set_client_side` (client_side)
	`set_control` (control)
	`set_dscp_qos` (dscp_qos)
	`set_max_mcast_ttl` (ttl)
	`set_mtu` (mtu)
	`set_multicast_iface` (multicast_iface)
	`set_profiles` (profiles)
	`set_protocols` (protocols)
	`set_pt_map` (pt, caps)
	`set_publish_clock_mode` (mode)
	`set_rate_control` (enabled)
	`set_retransmission_pt` (rtx_pt)
	`set_retransmission_time` (time)
	`set_seqnum_offset` (seqnum)
	`set_ulpfec_percentage` (percentage)
	`set_ulpfec_pt` (pt)
	`transport_filter` (func, *user_data)
	`unblock_linked` ()
	`update_crypto` (ssrc, crypto)
	`verify_mcast_ttl` (ttl)

Virtual Methods¶

Inherited:	GObject.Object (7)

Properties¶

Name	Type	Flags	Short Description
`control`	`str`	r/w	The control string for this stream
`profiles`	`GstRtsp.RTSPProfile`	r/w	Allowed transfer profiles
`protocols`	`GstRtsp.RTSPLowerTrans`	r/w	Allowed lower transport protocols

Signals¶

Inherited:	GObject.Object (1)

Name	Short Description
`new-rtcp-encoder`
`new-rtp-encoder`
`new-rtp-rtcp-decoder`

Fields¶

Inherited:	GObject.Object (1)

Name	Type	Access	Description
parent	`GObject.Object`	r

Class Details¶

class GstRtspServer.RTSPStream(**kwargs)¶

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

The definition of a media stream.

classmethod new(idx, payloader, pad)[source]¶

Parameters:	idx (`int`) – an index payloader (`Gst.Element`) – a `Gst.Element` pad (`Gst.Pad`) – a `Gst.Pad`
Returns:	a new `GstRtspServer.RTSPStream`
Return type:	`GstRtspServer.RTSPStream`

Create a new media stream with index idx that handles RTP GstRtsp.data on pad and has a payloader element payloader if pad is a source pad or a depayloader element payloader if pad is a sink pad.

add_multicast_client_address(destination, rtp_port, rtcp_port, family)[source]¶

Parameters:	destination (`str`) – a multicast address to add rtp_port (`int`) – RTP port rtcp_port (`int`) – RTCP port family (`Gio.SocketFamily`) – socket family
Returns:	`True` if destination can be addedd and handled by self.
Return type:	`bool`

Add multicast client address to stream. At this point, the sockets that will stream RTP and RTCP GstRtsp.data to destination are supposed to be allocated.

New in version 1.16.

add_transport(trans)[source]¶

Parameters:	trans (`GstRtspServer.RTSPStreamTransport`) – a `GstRtspServer.RTSPStreamTransport`
Returns:	`True` if trans was added
Return type:	`bool`

Add the transport in trans to self. The media of self will then also be send to the values configured in trans. Adding the same transport twice will not add it a second time.

self must be joined to a bin.

trans must contain a valid GstRtsp.RTSPTransport.

allocate_udp_sockets(family, transport, use_client_settings)[source]¶

Parameters:	family (`Gio.SocketFamily`) – protocol family transport (`GstRtsp.RTSPTransport`) – transport method use_client_settings (`bool`) – Whether to use client settings or not
Returns:	`True` if the RTP and RTCP sockets have been succeccully allocated.
Return type:	`bool`

Allocates RTP and RTCP ports.

complete_stream(transport)[source]¶

Parameters:	transport (`GstRtsp.RTSPTransport`) – a `GstRtsp.RTSPTransport`
Returns:	`True` if the stream has been sucessfully updated.
Return type:	`bool`

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

New in version 1.14.

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_buffer_size()[source]¶

Returns:	the size of the UDP TX buffer
Return type:	`int`

Get the size of the UDP transmission buffer (in bytes)

New in version 1.6.

get_caps()[source]¶

Returns:	the `Gst.Caps` of self. use gst_caps_unref() after usage.
Return type:	`Gst.Caps` or `None`

Retrieve the current caps of self.

get_control()[source]¶

Returns:	the control string. `GLib.free`() after usage.
Return type:	`str` or `None`

Get the control string to identify this stream.

get_current_seqnum()[source]¶

Return type:	`int`

get_dscp_qos()[source]¶

Returns:	the DSCP QoS value of the outgoing sockets, or -1 if disbled.
Return type:	`int`

Get the configured DSCP QoS in of the outgoing sockets.

get_index()[source]¶

Returns:	the stream index.
Return type:	`int`

Get the stream index.

get_joined_bin()[source]¶

Returns:	the joined bin or `None`.
Return type:	`Gst.Bin` or `None`

Get the previous joined bin with GstRtspServer.RTSPStream.join_bin() or None.

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_mtu()[source]¶

Returns:	the MTU of the payloader.
Return type:	`int`

Get the configured MTU in the payloader of self.

get_multicast_address(family)[source]¶

Parameters:	family (`Gio.SocketFamily`) – the `Gio.SocketFamily`
Returns:	the `GstRtspServer.RTSPAddress` of self or `None` when no address could be allocated. `GstRtspServer.RTSPAddress.free`() after usage.
Return type:	`GstRtspServer.RTSPAddress` or `None`

Get the multicast address of self for family. The original GstRtspServer.RTSPAddress is cached and copy is returned, so freeing the return value won’t release the address from the pool.

get_multicast_client_addresses()[source]¶

Returns:	A comma separated list of host:port pairs with destinations
Return type:	`str`

Get all multicast client addresses that RTP GstRtsp.data will be sent to

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_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_pt()[source]¶

Returns:	the stream payload type.
Return type:	`int`

Get the stream payload type.

get_publish_clock_mode()[source]¶

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

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

New in version 1.8.

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:

True if rate and/or applied_rate could be determined.

rate:	the configured rate
applied_rate:	the configured applied_rate

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

Retrieve the current rate and/or applied_rate.

New in version 1.18.

get_retransmission_pt()[source]¶

Returns:	The retransmission PT.
Return type:	`int`

Get the payload-type used for retransmission of this stream

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_rtcp_multicast_socket(family)[source]¶

Parameters:	family (`Gio.SocketFamily`) – the socket family
Returns:	the multicast RTCP socket or `None` if no socket could be allocated for family. Unref after usage
Return type:	`Gio.Socket` or `None`

Get the multicast RTCP socket from self for a family.

New in version 1.14.

get_rtcp_socket(family)[source]¶

Parameters:	family (`Gio.SocketFamily`) – the socket family
Returns:	the RTCP socket or `None` if no socket could be allocated for family. Unref after usage
Return type:	`Gio.Socket` or `None`

Get the RTCP socket from self for a family.

self must be joined to a bin.

get_rtp_multicast_socket(family)[source]¶

Parameters:	family (`Gio.SocketFamily`) – the socket family
Returns:	the multicast RTP socket or `None` if no socket could be allocated for family. Unref after usage
Return type:	`Gio.Socket` or `None`

Get the multicast RTP socket from self for a family.

get_rtp_socket(family)[source]¶

Parameters:	family (`Gio.SocketFamily`) – the socket family
Returns:	the RTP socket or `None` if no socket could be allocated for family. Unref after usage
Return type:	`Gio.Socket` or `None`

Get the RTP socket from self for a family.

self must be joined to a bin.

get_rtpinfo()[source]¶

Returns:

True when rtptime, seq and running-time could be determined.

rtptime:	result RTP timestamp
seq:	result RTP seqnum
clock_rate:	the clock rate
running_time:	result running-time

Return type: (bool, rtptime: int, seq: int, clock_rate: int, running_time: int)

Retrieve the current rtptime, seq and running-time. This is used to construct a RTPInfo reply header.

get_rtpsession()[source]¶

Returns:	The RTP session of this stream. Unref after usage.
Return type:	`GObject.Object`

Get the RTP session of this stream.

get_server_port(family)[source]¶

Parameters:	family (`Gio.SocketFamily`) – the port family to get
Returns:	result server port
Return type:	server_port: `GstRtsp.RTSPRange`

Fill server_port with the port pair used by the server. This function can only be called when self has been joined.

get_sinkpad()[source]¶

Returns:	the sinkpad. Unref after usage.
Return type:	`Gst.Pad` or `None`

Get the sinkpad associated with self.

get_srcpad()[source]¶

Returns:	the srcpad. Unref after usage.
Return type:	`Gst.Pad` or `None`

Get the srcpad associated with self.

get_srtp_encoder()[source]¶

Returns:	The SRTP encoder for this stream. Unref after usage.
Return type:	`Gst.Element`

Get the SRTP encoder for this stream.

get_ssrc()[source]¶

Returns:	result ssrc
Return type:	ssrc: `int`

Get the SSRC used by the RTP session of this stream. This function can only be called when self has been joined.

get_ulpfec_enabled()¶

Return type:	`bool`

get_ulpfec_percentage()[source]¶

Returns:	the amount of redundancy applied when creating ULPFEC protection packets.
Return type:	`int`

New in version 1.16.

get_ulpfec_pt()[source]¶

Returns:	the payload type used for ULPFEC protection packets
Return type:	`int`

New in version 1.16.

handle_keymgmt(keymgmt)[source]¶

Parameters:	keymgmt (`str`) – a keymgmt header
Return type:	`bool`

Parse and handle a KeyMgmt header.

New in version 1.16.

has_control(control)[source]¶

Parameters:	control (`str` or `None`) – a control string
Returns:	`True` is self has control as the control string
Return type:	`bool`

Check if self has the control string control.

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_blocking()[source]¶

Returns:	`True` if self is blocking
Return type:	`bool`

Check if self is blocking on a Gst.Buffer.

is_client_side()[source]¶

Returns:	`True` if this `GstRtspServer.RTSPStream` is client-side.
Return type:	`bool`

See GstRtspServer.RTSPStream.set_client_side()

is_complete()[source]¶

Returns:	`True` if the stream contains at least one sink element.
Return type:	`bool`

Checks whether the stream is complete, contains the receiver and the sender parts. As the stream contains sink(s) element(s), it’s possible to perform seek operations on it.

New in version 1.14.

is_receiver()[source]¶

Returns:	`True` if the stream is a receiver and `False` otherwise.
Return type:	`bool`

Checks whether the stream is a receiver.

New in version 1.14.

is_sender()[source]¶

Returns:	`True` if the stream is a sender and `False` otherwise.
Return type:	`bool`

Checks whether the stream is a sender.

New in version 1.14.

is_transport_supported(transport)[source]¶

Parameters:	transport (`GstRtsp.RTSPTransport`) – a `GstRtsp.RTSPTransport`
Returns:	`True` if transport can be handled by self.
Return type:	`bool`

Check if transport can be handled by stream

join_bin(bin, rtpbin, state)[source]¶

Parameters:	bin (`Gst.Bin`) – a `Gst.Bin` to join rtpbin (`Gst.Element`) – a rtpbin element in bin state (`Gst.State`) – the target state of the new elements
Returns:	`True` on success.
Return type:	`bool`

Join the Gst.Bin bin that contains the element rtpbin.

self will link to rtpbin, which must be inside bin. The elements added to bin will be set to the state given in state.

leave_bin(bin, rtpbin)[source]¶

Parameters:	bin (`Gst.Bin`) – a `Gst.Bin` rtpbin (`Gst.Element`) – a rtpbin `Gst.Element`
Returns:	`True` on success.
Return type:	`bool`

Remove the elements of self from bin.

query_position()[source]¶

Returns:

True if the position could be queried

position:	current position of a `GstRtspServer.RTSPStream`

Return type: (bool, position: int)

Query the position of the stream in Gst.Format.TIME. This only considers the RTP parts of the pipeline and not the RTCP parts.

query_stop()[source]¶

Returns:

True if the stop could be queried

stop:	current stop of a `GstRtspServer.RTSPStream`

Return type: (bool, stop: int)

Query the stop of the stream in Gst.Format.TIME. This only considers the RTP parts of the pipeline and not the RTCP parts.

recv_rtcp(buffer)[source]¶

Parameters:	buffer (`Gst.Buffer`) – a `Gst.Buffer`
Returns:	a `Gst.FlowReturn`.
Return type:	`Gst.FlowReturn`

Handle an RTCP buffer for the stream. This method is usually called when a message has been received from a client using the TCP transport.

This function takes ownership of buffer.

recv_rtp(buffer)[source]¶

Parameters:	buffer (`Gst.Buffer`) – a `Gst.Buffer`
Returns:	a `Gst.FlowReturn`.
Return type:	`Gst.FlowReturn`

Handle an RTP buffer for the stream. This method is usually called when a message has been received from a client using the TCP transport.

This function takes ownership of buffer.

remove_transport(trans)[source]¶

Parameters:	trans (`GstRtspServer.RTSPStreamTransport`) – a `GstRtspServer.RTSPStreamTransport`
Returns:	`True` if trans was removed
Return type:	`bool`

Remove the transport in trans from self. The media of self will not be sent to the values configured in trans.

self must be joined to a bin.

trans must contain a valid GstRtsp.RTSPTransport.

request_aux_receiver(sessid)[source]¶

Parameters:	sessid (`int`) – the session id
Returns:	a `Gst.Element`.
Return type:	`Gst.Element` or `None`

Creating a rtxreceive bin

New in version 1.16.

request_aux_sender(sessid)[source]¶

Parameters:	sessid (`int`) – the session id
Returns:	a `Gst.Element`.
Return type:	`Gst.Element` or `None`

Creating a rtxsend bin

New in version 1.6.

request_ulpfec_decoder(rtpbin, sessid)[source]¶

Parameters:	rtpbin (`Gst.Element`) – sessid (`int`) –
Returns:	a `Gst.Element`.
Return type:	`Gst.Element` or `None`

Creating a rtpulpfecdec element

New in version 1.16.

request_ulpfec_encoder(sessid)[source]¶

Parameters:	sessid (`int`) –
Returns:	a `Gst.Element`.
Return type:	`Gst.Element` or `None`

Creating a rtpulpfecenc element

New in version 1.16.

reserve_address(address, port, n_ports, ttl)[source]¶

Parameters:	address (`str`) – an address port (`int`) – a port n_ports (`int`) – n_ports ttl (`int`) – a TTL
Returns:	the `GstRtspServer.RTSPAddress` of self or `None` when the address could not be reserved. `GstRtspServer.RTSPAddress.free`() after usage.
Return type:	`GstRtspServer.RTSPAddress` or `None`

Reserve address and port as the address and port of self. The original GstRtspServer.RTSPAddress is cached and copy is returned, so freeing the return value won’t release the address from the pool.

seekable()[source]¶

Returns:	`True` if self is seekable, else `False`.
Return type:	`bool`

Checks whether the individual self is seekable.

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_blocked(blocked)[source]¶

Parameters:	blocked (`bool`) – boolean indicating we should block or unblock
Returns:	`True` on success
Return type:	`bool`

Blocks or unblocks the dataflow on self.

set_buffer_size(size)[source]¶

Parameters:	size (`int`) – the buffer size

Set the size of the UDP transmission buffer (in bytes) Needs to be set before the stream is joined to a bin.

New in version 1.6.

set_client_side(client_side)[source]¶

Parameters:	client_side (`bool`) – `True` if this `GstRtspServer.RTSPStream` is running on the ‘client’ side of an RTSP connection.

Sets the GstRtspServer.RTSPStream as a ‘client side’ stream - used for sending streams to an RTSP server via RECORD. This has the practical effect of changing which UDP port numbers are used when setting up the local side of the stream sending to be either the ‘server’ or ‘client’ pair of a configured UDP transport.

set_control(control)[source]¶

Parameters:	control (`str` or `None`) – a control string

Set the control string in self.

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 the outgoing sockets to dscp_qos.

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_mtu(mtu)[source]¶

Parameters:	mtu (`int`) – a new MTU

Configure the mtu in the payloader of self to mtu.

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_profiles(profiles)[source]¶

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

Configure the allowed profiles for self.

set_protocols(protocols)[source]¶

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

Configure the allowed lower transport for self.

set_pt_map(pt, caps)[source]¶

Parameters:	pt (`int`) – the pt caps (`Gst.Caps`) – a `Gst.Caps`

Configure a pt map between pt and caps.

set_publish_clock_mode(mode)[source]¶

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

Sets if and how the stream 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_pt(rtx_pt)[source]¶

Parameters:	rtx_pt (`int`) – a `int`

Set the payload type (pt) for retransmission of this stream.

set_retransmission_time(time)[source]¶

Parameters:	time (`int`) – a #GstClockTime

Set the amount of time to store retransmission packets.

set_seqnum_offset(seqnum)[source]¶

Parameters:	seqnum (`int`) –

set_ulpfec_percentage(percentage)[source]¶

Parameters:	percentage (`int`) –

Sets the amount of redundancy to apply when creating ULPFEC protection packets.

New in version 1.16.

set_ulpfec_pt(pt)[source]¶

Parameters:	pt (`int`) –

Set the payload type to be used for ULPFEC protection packets

New in version 1.16.

transport_filter(func, *user_data)[source]¶

Parameters:	func (`GstRtspServer.RTSPStreamTransportFilterFunc` or `None`) – a callback user_data (`object` or `None`) – user `GstRtsp.data` passed to func
Returns:	a `GLib.List` with all transports for which func returned `GstRtspServer.RTSPFilterResult.REF`. After usage, each element in the `GLib.List` should be unreffed before the list is freed.
Return type:	[`GstRtspServer.RTSPStreamTransport`]

Call func for each transport managed by self. The result value of func determines what happens to the transport. func will be called with self locked so no further actions on self can be performed from func.

If func returns GstRtspServer.RTSPFilterResult.REMOVE, the transport will be removed from self.

If func returns GstRtspServer.RTSPFilterResult.KEEP, the transport will remain in self.

If func returns GstRtspServer.RTSPFilterResult.REF, the transport will remain in self but will also be added with an additional ref to the result GLib.List of this function..

When func is None, GstRtspServer.RTSPFilterResult.REF will be assumed for each transport.

unblock_linked()[source]¶

Return type:	`bool`

update_crypto(ssrc, crypto)[source]¶

Parameters:	ssrc (`int`) – the SSRC crypto (`Gst.Caps` or `None`) – a `Gst.Caps` with crypto info
Returns:	`True` if crypto could be updated
Return type:	`bool`

Update the new crypto information for ssrc in self. If information for ssrc did not exist, it will be added. If information for ssrc existed, it will be replaced. If crypto is None, it will be removed from self.

verify_mcast_ttl(ttl)[source]¶

Parameters:	ttl (`int`) – a requested multicast ttl
Returns:	`True` if the requested ttl value is allowed.
Return type:	`bool`

Check if the requested multicast ttl value is allowed.

New in version 1.16.

Signal Details¶

GstRtspServer.RTSPStream.signals.new_rtcp_encoder(r_t_s_p_stream, object)¶

Signal Name:	`new-rtcp-encoder`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_stream (`GstRtspServer.RTSPStream`) – The object which received the signal object (`Gst.Element`) –

GstRtspServer.RTSPStream.signals.new_rtp_encoder(r_t_s_p_stream, object)¶

Signal Name:	`new-rtp-encoder`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_stream (`GstRtspServer.RTSPStream`) – The object which received the signal object (`Gst.Element`) –

GstRtspServer.RTSPStream.signals.new_rtp_rtcp_decoder(r_t_s_p_stream, object)¶

Signal Name:	`new-rtp-rtcp-decoder`
Flags:	`RUN_LAST`
Parameters:	r_t_s_p_stream (`GstRtspServer.RTSPStream`) – The object which received the signal object (`Gst.Element`) –

Property Details¶

GstRtspServer.RTSPStream.props.control¶

Name:	`control`
Type:	`str`
Default Value:	`None`
Flags:	`READABLE`, `WRITABLE`

The control string for this stream

GstRtspServer.RTSPStream.props.profiles¶

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

Allowed transfer profiles

GstRtspServer.RTSPStream.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