rtspsrc

Hierarchy

GObject
    ╰──GInitiallyUnowned
        ╰──GstObject
            ╰──GstElement
                ╰──GstBin
                    ╰──rtspsrc

Implemented interfaces

GstChildProxy
GstURIHandler

Factory details

stream_%u

application/x-rtp:
application/x-rdt:

accept-certificate

gboolean
accept_certificate_callback (GstElement * rtspsrc,
                             GTlsConnection * peer_cert,
                             GTlsCertificate * errors,
                             GTlsCertificateFlags * user_data,
                             gpointer udata)

def accept_certificate_callback (rtspsrc, peer_cert, errors, user_data, udata):
    #python callback for the 'accept-certificate' signal

function accept_certificate_callback(rtspsrc: GstElement * rtspsrc, peer_cert: GTlsConnection * peer_cert, errors: GTlsCertificate * errors, user_data: GTlsCertificateFlags * user_data, udata: gpointer udata): {
    // javascript callback for the 'accept-certificate' signal
}

This will directly map to GTlsConnection 's "accept-certificate" signal and be performed after the default checks of GstRTSPConnection (checking against the GTlsDatabase with the given GTlsCertificateFlags) have failed. If no GTlsDatabase is set on this connection, only this signal will be emitted.

Flags: Run Last

Since : 1.14

before-send

gboolean
before_send_callback (GstElement * rtspsrc,
                      GstRTSPMessage * num,
                      gpointer udata)

def before_send_callback (rtspsrc, num, udata):
    #python callback for the 'before-send' signal

function before_send_callback(rtspsrc: GstElement * rtspsrc, num: GstRTSPMessage * num, udata: gpointer udata): {
    // javascript callback for the 'before-send' signal
}

Emitted before each RTSP request is sent, in order to allow the application to modify send parameters or to skip the message entirely. This can be used, for example, to work with ONVIF Profile G servers, which need a different/additional range, rate-control, and intra/x parameters.

Flags: Run Last

Since : 1.14

handle-request

handle_request_callback (GstElement * rtspsrc,
                         GstRTSPMessage * request,
                         GstRTSPMessage * response,
                         gpointer udata)

def handle_request_callback (rtspsrc, request, response, udata):
    #python callback for the 'handle-request' signal

function handle_request_callback(rtspsrc: GstElement * rtspsrc, request: GstRTSPMessage * request, response: GstRTSPMessage * response, udata: gpointer udata): {
    // javascript callback for the 'handle-request' signal
}

Handle a server request in request and prepare response.

This signal is called from the streaming thread, you should therefore not do any state changes on rtspsrc because this might deadlock. If you want to modify the state as a result of this signal, post a GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread in some other way.

Flags:

Since : 1.2

hard-limit

hard_limit_callback (GstElement * rtspsrc,
                     guint id,
                     gpointer udata)

def hard_limit_callback (rtspsrc, id, udata):
    #python callback for the 'hard-limit' signal

function hard_limit_callback(rtspsrc: GstElement * rtspsrc, id: guint id, udata: gpointer udata): {
    // javascript callback for the 'hard-limit' signal
}

When the 'client-managed-mikey' mode is enabled, this signal is emitted when the stream with id has reached the hard limit of utilisation of it's master encryption key. User should start a re-keying procedure. Failure to do so, will lead to buffers of this stream being dropped.

Flags: Run Last

Since : 1.26

new-manager

new_manager_callback (GstElement * rtspsrc,
                      GstElement * manager,
                      gpointer udata)

def new_manager_callback (rtspsrc, manager, udata):
    #python callback for the 'new-manager' signal

function new_manager_callback(rtspsrc: GstElement * rtspsrc, manager: GstElement * manager, udata: gpointer udata): {
    // javascript callback for the 'new-manager' signal
}

Emitted after a new manager (like rtpbin) was created and the default properties were configured.

Flags: Run First

Since : 1.4

on-sdp

on_sdp_callback (GstElement * rtspsrc,
                 GstSDPMessage * sdp,
                 gpointer udata)

def on_sdp_callback (rtspsrc, sdp, udata):
    #python callback for the 'on-sdp' signal

function on_sdp_callback(rtspsrc: GstElement * rtspsrc, sdp: GstSDPMessage * sdp, udata: gpointer udata): {
    // javascript callback for the 'on-sdp' signal
}

Emitted when the client has retrieved the SDP and before it configures the streams in the SDP. sdp can be inspected and modified.

This signal is called from the streaming thread, you should therefore not do any state changes on rtspsrc because this might deadlock. If you want to modify the state as a result of this signal, post a GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread in some other way.

Flags:

Since : 1.2

request-rtcp-key

GstCaps *
request_rtcp_key_callback (GstElement * rtspsrc,
                           guint num,
                           gpointer udata)

def request_rtcp_key_callback (rtspsrc, num, udata):
    #python callback for the 'request-rtcp-key' signal

function request_rtcp_key_callback(rtspsrc: GstElement * rtspsrc, num: guint num, udata: gpointer udata): {
    // javascript callback for the 'request-rtcp-key' signal
}

Signal emitted to get the crypto parameters relevant to the RTCP stream. User should provide the key and the RTCP encryption ciphers and authentication, and return them wrapped in a GstCaps.

Flags: Run Last

Since : 1.4

request-rtp-key

GstCaps *
request_rtp_key_callback (GstElement * rtspsrc,
                          guint num,
                          gpointer udata)

def request_rtp_key_callback (rtspsrc, num, udata):
    #python callback for the 'request-rtp-key' signal

function request_rtp_key_callback(rtspsrc: GstElement * rtspsrc, num: guint num, udata: gpointer udata): {
    // javascript callback for the 'request-rtp-key' signal
}

Signal emitted to get the crypto parameters relevant to the RTP stream. User should provide the key and the RTP encryption ciphers and authentication, and return them wrapped in a GstCaps.

Applications should connect to this signal when 'client-managed-mikey' mode is enabled. The crypto parameters are usually the same as those use by the 'request-rtcp-key' signal.

Flags:

Since : 1.26

select-stream

gboolean
select_stream_callback (GstElement * rtspsrc,
                        guint num,
                        GstCaps * caps,
                        gpointer udata)

def select_stream_callback (rtspsrc, num, caps, udata):
    #python callback for the 'select-stream' signal

function select_stream_callback(rtspsrc: GstElement * rtspsrc, num: guint num, caps: GstCaps * caps, udata: gpointer udata): {
    // javascript callback for the 'select-stream' signal
}

Emitted before the client decides to configure the stream num with caps.

Flags: Run Last

Since : 1.2

soft-limit

soft_limit_callback (GstElement * rtspsrc,
                     guint id,
                     gpointer udata)

def soft_limit_callback (rtspsrc, id, udata):
    #python callback for the 'soft-limit' signal

function soft_limit_callback(rtspsrc: GstElement * rtspsrc, id: guint id, udata: gpointer udata): {
    // javascript callback for the 'soft-limit' signal
}

When the 'client-managed-mikey' mode is enabled, this signal is emitted when the stream with id has reached the soft limit of utilisation of it's master encryption key. User should start a re-keying procedure.

Flags: Run Last

Since : 1.26

get-parameter

g_signal_emit_by_name (rtspsrc, "get-parameter", parameter, arg1, arg2, &ret);

ret = rtspsrc.emit ("get-parameter", parameter, arg1, arg2)

let ret = rtspsrc.emit ("get-parameter", parameter, arg1, arg2);

Handle the GET_PARAMETER signal.

Flags: Run Last / Action

get-parameters

g_signal_emit_by_name (rtspsrc, "get-parameters", parameter, arg1, arg2, &ret);

ret = rtspsrc.emit ("get-parameters", parameter, arg1, arg2)

let ret = rtspsrc.emit ("get-parameters", parameter, arg1, arg2);

Handle the GET_PARAMETERS signal.

Flags: Run Last / Action

push-backchannel-buffer

g_signal_emit_by_name (rtspsrc, "push-backchannel-buffer", id, sample, &ret);

ret = rtspsrc.emit ("push-backchannel-buffer", id, sample)

let ret = rtspsrc.emit ("push-backchannel-buffer", id, sample);

Flags: Run Last / Action

deprecated :

1.22: Use action signal GstRTSPSrc::push-backchannel-sample instead. IMPORTANT: Please note that this signal decrements the reference count of sample internally! So it cannot be used from other language bindings in general.

push-backchannel-sample

g_signal_emit_by_name (rtspsrc, "push-backchannel-sample", id, sample, &ret);

ret = rtspsrc.emit ("push-backchannel-sample", id, sample)

let ret = rtspsrc.emit ("push-backchannel-sample", id, sample);

Flags: Run Last / Action

Since : 1.22

remove-key

g_signal_emit_by_name (rtspsrc, "remove-key", parameter, &ret);

ret = rtspsrc.emit ("remove-key", parameter)

let ret = rtspsrc.emit ("remove-key", parameter);

Removes the key for a specific stream.

When the 'client-managed-mikey' mode is enabled, this can be used after informing the server of the new crypto params (see signal 'set-mikey-parameter') to remove previous keys and force srtpdec to request new keys.

Flags: Run Last / Action

Since : 1.26

set-mikey-parameter

g_signal_emit_by_name (rtspsrc, "set-mikey-parameter", parameter, arg1, arg2, &ret);

ret = rtspsrc.emit ("set-mikey-parameter", parameter, arg1, arg2)

let ret = rtspsrc.emit ("set-mikey-parameter", parameter, arg1, arg2);

Sends a SET_PARAMETER to the server with a MIKEY payload.

This will call SET_PARAMETER with parameter 'KeyMgmt' and a MIKEY payload formed from the provided stream's ssrc and MIKEY caps.

The GstPromise reply consists in the following fields:

• 'rtsp-result': set to 0 if the HTTP request could be processed.
• 'rtsp-code': the HTTP status code returned by the server.
• 'rtsp-reason': a human-readable version of the HTTP status code.

Flags: Run Last / Action

Since : 1.26

set-parameter

g_signal_emit_by_name (rtspsrc, "set-parameter", parameter, arg1, arg2, arg3, &ret);

ret = rtspsrc.emit ("set-parameter", parameter, arg1, arg2, arg3)

let ret = rtspsrc.emit ("set-parameter", parameter, arg1, arg2, arg3);

Handle the SET_PARAMETER signal.

The GstPromise reply consists in the following fields:

• 'rtsp-result': set to 0 if the HTTP request could be processed.
• 'rtsp-code': the HTTP status code returned by the server.
• 'rtsp-reason': a human-readable version of the HTTP status code.

Flags: Run Last / Action

add-reference-timestamp-meta

“add-reference-timestamp-meta” gboolean

When syncing to a RFC7273 clock, add GstReferenceTimestampMeta to buffers with the original reconstructed reference clock timestamp.

Flags : Read / Write

Default value : false

Since : 1.22

backchannel

“backchannel” Rtspbackchannel *

Select a type of backchannel to setup with the RTSP server. Default value is "none". Allowed values are "none" and "onvif".

Flags : Read / Write

Default value : none (0)

Since : 1.14

buffer-mode

“buffer-mode” Rtspsrc-buffer-mode *

Control the buffering and timestamping mode used by the jitterbuffer.

Flags : Read / Write

Default value : auto (3)

client-managed-mikey

“client-managed-mikey” gboolean

Some servers (e.g. Axis Cameras) advertise a key-mgmt:mikey attribute in the SDP from the reply to DESCRIBE, but expect the client to provide the key for both the client (SRTCP/RR) and the server (SRTP/SRTCP SR).

When this mode is enabled, for each eligible media, the server crypto policy is discarded. The crypto policy for this media is built from the key returned by the signals 'request-rtp-key' & 'request-rtcp-key'.

A media is eligible for this mode if:

• No session level MIKEY key-mgmt are defined and the media level key-mgmt is MIKEY.
• A session level MIKEY key-mgmt is defined and no media level key-mgmts are defined.
• A session level MIKEY key-mgmt is defined and the media level key-mgmt is also MIKEY.

Flags : Read / Write

Default value : false

Since : 1.26

connection-speed

“connection-speed” guint64

Network connection speed in kbps (0 = unknown)

Flags : Read / Write

Default value : 0

debug

“debug” gboolean

Dump request and response messages to stdout(DEPRECATED: Printed all RTSP message to gstreamer log as 'log' level)

Flags : Read / Write

Default value : false

default-rtsp-version

“default-rtsp-version” GstRTSPVersion *

The preferred RTSP version to use while negotiating the version with the server.

Flags : Read / Write

Default value : 1-0 (16)

Since : 1.14

do-retransmission

“do-retransmission” gboolean

Ask the server to retransmit lost packets

Flags : Read / Write

Default value : true

do-rtcp

“do-rtcp” gboolean

Enable RTCP support. Some old server don't like RTCP and then this property needs to be set to FALSE.

Flags : Read / Write

Default value : true

do-rtsp-keep-alive

“do-rtsp-keep-alive” gboolean

Enable RTSP keep alive support. Some old server don't like RTSP keep alive and then this property needs to be set to FALSE.

Flags : Read / Write

Default value : true

drop-on-latency

“drop-on-latency” gboolean

Tells the jitterbuffer to never exceed the given latency in size

Flags : Read / Write

Default value : false

extra-http-request-headers

“extra-http-request-headers” GstStructure *

When in tunneled mode append provided headers to any HTTP requests made by rtspsrc.

Only applicable for RTSP over HTTP.

Flags : Read / Write

Default value :

extra-http-request-headers;

Since : 1.24

force-non-compliant-url

“force-non-compliant-url” gboolean

There are various non-compliant servers that don't require control URLs that are not resolved correctly but instead are just appended.

As some of these servers will nevertheless reply OK to SETUP requests even if they didn't handle URIs correctly, this property can be set to revert to the old non-compliant method for constructing URLs.

Flags : Read / Write

Default value : false

Since : 1.24.7

ignore-x-server-reply

“ignore-x-server-reply” gboolean

When connecting to an RTSP server in tunneled mode (HTTP) the server usually replies with an x-server-ip-address header. This contains the address of the intended streaming server. However some servers return an "invalid" address. Here follows two examples when it might happen.

1. A server uses Apache combined with a separate RTSP process to handle HTTPS requests on port 443. In this case Apache handles TLS and connects to the local RTSP server, which results in a local address 127.0.0.1 or ::1 in the header reply. This address is returned to the actual RTSP client in the header. The client will receive this address and try to connect to it and fail.

2. The client uses an IPv6 link local address with a specified scope id fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via HTTP on port 80. The RTSP server receives the connection and returns the address in the x-server-ip-address header. The client will receive this address and try to connect to it "as is" without the scope id and fail.

In the case of streaming data from RTSP servers like 1 and 2, it's useful to have the option to simply ignore the x-server-ip-address header reply and continue using the original address.

Flags : Read / Write

Default value : false

Since : 1.20

is-live

“is-live” gboolean

Whether to act as a live source. This is useful in combination with onvif-rate-control set to FALSE and usage of the TCP protocol. In that situation, data delivery rate can be entirely controlled from the client side, enabling features such as frame stepping and instantaneous rate changes.

Flags : Read / Write

Default value : true

Since : 1.18

latency

“latency” guint

Amount of ms to buffer

Flags : Read / Write

Default value : 2000

location

“location” gchararray

Location of the RTSP url to read

Flags : Read / Write

Default value : NULL

max-rtcp-rtp-time-diff

“max-rtcp-rtp-time-diff” gint

Maximum amount of time in ms that the RTP time in RTCP SRs is allowed to be ahead (-1 disabled)

Flags : Read / Write

Default value : -1

max-ts-offset

“max-ts-offset” gint64

Used to set an upper limit of how large a time offset may be. This is used to protect against unrealistic values as a result of either client,server or clock issues.

Flags : Read / Write

Default value : 3000000000

max-ts-offset-adjustment

“max-ts-offset-adjustment” guint64

Syncing time stamps to NTP time adds a time offset. This parameter specifies the maximum number of nanoseconds per frame that this time offset may be adjusted with. This is used to avoid sudden large changes to time stamps.

Flags : Read / Write

Default value : 0

multicast-iface

“multicast-iface” gchararray

The network interface on which to join the multicast group

Flags : Read / Write

Default value : NULL

nat-method

“nat-method” Rtspnat-method *

Method to use for traversing firewalls and NAT

Flags : Read / Write

Default value : dummy (1)

ntp-sync

“ntp-sync” gboolean

Synchronize received streams to the NTP clock

Flags : Read / Write

Default value : false

ntp-time-source

“ntp-time-source” Rtspsrc-ntp-time-source *

NTP time source for RTCP packets

Flags : Read / Write

Default value : ntp (0)

onvif-mode

“onvif-mode” gboolean

Act as an ONVIF client. When set to TRUE:

• seeks will be interpreted as nanoseconds since prime epoch (1900-01-01)

• onvif-rate-control can be used to request that the server sends data as fast as it can

• TCP is picked as the transport protocol

• Trickmode flags in seek events are transformed into the appropriate ONVIF request headers

Flags : Read / Write

Default value : false

Since : 1.18

onvif-rate-control

“onvif-rate-control” gboolean

When in onvif-mode, whether to set Rate-Control to yes or no. When set to FALSE, the server will deliver data as fast as the client can consume it.

Flags : Read / Write

Default value : true

Since : 1.18

port-range

“port-range” gchararray

Configure the client port numbers that can be used to receive RTP and RTCP.

Flags : Read / Write

Default value : NULL

probation

“probation” guint

Consecutive packet sequence numbers to accept the source

Flags : Read / Write

Default value : 2

protocols

“protocols” GstRTSPLowerTrans *

Allowed lower transport protocols

Flags : Read / Write

Default value : tcp+udp-mcast+udp

proxy

“proxy” gchararray

Set the proxy parameters. This has to be a string of the format [http://][user:passwd@]host[:port].

Flags : Read / Write

Default value : NULL

proxy-id

“proxy-id” gchararray

Sets the proxy URI user id for authentication. If the URI set via the "proxy" property contains a user-id already, that will take precedence.

Flags : Read / Write

Default value : NULL

Since : 1.2

proxy-pw

“proxy-pw” gchararray

Sets the proxy URI password for authentication. If the URI set via the "proxy" property contains a password already, that will take precedence.

Flags : Read / Write

Default value : NULL

Since : 1.2

retry

“retry” guint

Max number of retries when allocating RTP ports.

Flags : Read / Write

Default value : 20

rfc7273-sync

“rfc7273-sync” gboolean

Synchronize received streams to the RFC7273 clock (requires clock and offset to be provided)

Flags : Read / Write

Default value : false

rtp-blocksize

“rtp-blocksize” guint

RTP package size to suggest to server.

Flags : Read / Write

Default value : 0

sdes

“sdes” GstStructure *

The SDES items of this session

Flags : Read / Write

short-header

“short-header” gboolean

Only send the basic RTSP headers for broken encoders.

Flags : Read / Write

Default value : false

tcp-timeout

“tcp-timeout” guint64

Fail after timeout microseconds on TCP connections (0 = disabled)

Flags : Read / Write

Default value : 20000000

tcp-timestamp

“tcp-timestamp” gboolean

Timestamp all buffers with their receive time when receiving RTP packets over TCP or HTTP.

When dealing with TCP based connections, setting timestamps for every packet is not done by default because a server typically bursts data, for which we don't want to compensate by speeding up the media. The other timestamps will be interpollated from this one using the RTP timestamps.

This has the side effect that no drift compensation between the server and client is done, and over time the RTP timestamps will drift against the client's clock. This can lead to buffers (and observed end-to-end latency) to grow over time, or all packets arriving too late once a threshold is reached.

Enabling this property will timestamp all RTP packets with their receive times, which gets around the drift problem but can cause other problems if the server is sending data not smoothly in real-time.

Only applicable for RTSP over TCP or HTTP.

Flags : Read / Write

Default value : false

Since : 1.26

teardown-timeout

“teardown-timeout” guint64

When transitioning PAUSED-READY, allow up to timeout (in nanoseconds) delay in order to send teardown (0 = disabled)

Flags : Read / Write

Default value : 100000000

Since : 1.14

timeout

“timeout” guint64

Retry TCP transport after UDP timeout microseconds (0 = disabled)

Flags : Read / Write

Default value : 5000000

tls-database

“tls-database” GTlsDatabase *

TLS database with anchor certificate authorities used to validate the server certificate

Flags : Read / Write

tls-interaction

“tls-interaction” GTlsInteraction *

A GTlsInteraction object to prompt the user for password or certificate

Flags : Read / Write

tls-validation-flags

“tls-validation-flags” GTlsCertificateFlags *

TLS certificate validation flags used to validate the server certificate

Flags : Read / Write

Default value : validate-all

udp-buffer-size

“udp-buffer-size” gint

Size of the kernel UDP receive buffer in bytes.

Flags : Read / Write

Default value : 524288

udp-reconnect

“udp-reconnect” gboolean

Reconnect to the server if RTSP connection is closed when doing UDP

Flags : Read / Write

Default value : true

use-pipeline-clock

“use-pipeline-clock” gboolean

Use the pipeline running-time to set the NTP time in the RTCP SR messages(DEPRECATED: Use ntp-time-source property)

Flags : Read / Write

Default value : false

user-agent

“user-agent” gchararray

The User-Agent string to send to the server

Flags : Read / Write

Default value : GStreamer/{VERSION}

user-id

“user-id” gchararray

RTSP location URI user id for authentication

Flags : Read / Write

Default value : NULL

user-pw

“user-pw” gchararray

RTSP location URI user password for authentication

Flags : Read / Write

Default value : NULL

Rtspbackchannel

Rtspnat-method

Different methods for trying to traverse firewalls.

Rtspsrc-buffer-mode

Rtspsrc-ntp-time-source