# rtspsrc

Makes a connection to an RTSP server and read the data. rtspsrc strictly follows RFC 2326 and therefore does not (yet) support RealMedia/Quicktime/Microsoft extensions.

RTSP supports transport over TCP or UDP in unicast or multicast mode. By default rtspsrc will negotiate a connection in the following order: UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed protocols can be controlled with the protocols property.

rtspsrc currently understands SDP as the format of the session description. For each stream listed in the SDP a new rtp_stream%d pad will be created with caps derived from the SDP media description. This is a caps of mime type "application/x-rtp" that can be connected to any available RTP depayloader element.

rtspsrc will internally instantiate an RTP session manager element that will handle the RTCP messages to and from the server, jitter removal, packet reordering along with providing a clock for the pipeline. This feature is implemented using the gstrtpbin element.

rtspsrc acts like a live source and will therefore only generate data in the PLAYING state.

If a RTP session times out then the rtspsrc will generate an element message named "GstRTSPSrcTimeout". Currently this is only supported for timeouts triggered by RTCP.

The message's structure contains three fields:

GstRTSPSrcTimeoutCause `cause`: the cause of the timeout.

gint `stream-number`: an internal identifier of the stream that timed out.

guint `ssrc`: the SSRC of the stream that timed out.

## Example launch line

     gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink

Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.

NOTE: rtspsrc will send a PAUSE command to the server if you set the element to the PAUSED state, and will send a PLAY command if you set it to the PLAYING state.

Unfortunately, going to the NULL state involves going through PAUSED, so rtspsrc does not know the difference and will send a PAUSE when you wanted a TEARDOWN. The workaround is to hook into the `before-send` signal and return FALSE in this case.

Some servers (e.g. Axis cameras) expect the client to propose the encryption key(s) to be used for SRTP / SRTCP. This is required to allow re-keying so as to evade cryptanalysis. Note that the behaviour is not specified by the RFCs. By setting the 'client-managed-mikey-mode' property to 'true', rtspsrc acts as follows:

- For a secured profile (RTP/SAVP or RTP/SAVPF), any media in the SDP returned by the server for which MIKEY key management applies is elligible for client managed mode. The MIKEY from the server is then ignored.
- rtspsrc sends a SETUP with a MIKEY payload proposed by the user. The payload is formed by calling the 'request-rtp-key' signal for each elligible stream. During initialisation, 'request-rtcp-key' is also called as usual. The keys returned by both signals should be the same for a single stream, but the mechanism allows a different approach.
- The user can start re-keying of a stream by calling SET_PARAMETER. The convenience signal 'set-mikey-parameter' can be used to build a 'KeyMgmt' parameter with a MIKEY payload.
- After the server accepts the new parameter, the user can call 'remove-key' and prepare for the new key(s) to be served by signals 'request-rtp-key' & 'request-rtcp-key'.
- The signals 'soft-limit' & 'hard-limit' are called when a key reaches the limits of its utilisation.

## Hierarchy

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

## Implemented interfaces

    GstChildProxy
    GstURIHandler

## Factory details

Authors: – Wim Taymans , Thijs Vermeir , Lutz Mueller

Classification: – `Source/Network`

Rank – primary

Plugin – rtsp

Package – GStreamer Good Plug-ins

## Pad Templates

### `stream_%u`

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

Presence – sometimes

Direction – src

Object type – GstPad

## Signals

### 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.

Parameters:

`rtspsrc` –

a rtspsrc

`peer_cert` –

the peer's GTlsCertificate

`errors` –

the problems with peer_cert

`user_data` –

user data set when the signal handler was connected.

`udata` –
No description available
Returns (gboolean) –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`num` –

the stream number

`udata` –
No description available
Returns (gboolean) –

TRUE when the command should be sent, FALSE when the command should be dropped.

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.

Parameters:

`rtspsrc` –

a rtspsrc

`request` –

a GstRTSPMessage

`response` –

a GstRTSPMessage

`udata` –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`id` –

The id of the stream

`udata` –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`manager` –

a GstElement

`udata` –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`sdp` –

a GstSDPMessage

`udata` –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`num` –

the stream number

`udata` –
No description available
Returns (GstCaps *) –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`num` –

the stream number

`udata` –
No description available
Returns (GstCaps *) –
No description available

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.

Parameters:

`rtspsrc` –

a rtspsrc

`num` –

the stream number

`caps` –

the stream caps

`udata` –
No description available
Returns (gboolean) –

TRUE when the stream should be selected, FALSE when the stream is to be ignored.

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.

Parameters:

`rtspsrc` –

a rtspsrc

`id` –

The id of the stream

`udata` –
No description available

Flags: Run Last

Since : 1.26

## Action Signals

### 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.

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`parameter` (gchararray) –

a pointer to GstPromise

`arg1` (gchararray) –
No description available
`arg2` (GstPromise *) –
No description available
Returns (gboolean) –

TRUE when the command could be issued, FALSE otherwise

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.

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`parameter` (GStrv *) –

a pointer to GstPromise

`arg1` (gchararray) –
No description available
`arg2` (GstPromise *) –
No description available
Returns (gboolean) –

TRUE when the command could be issued, FALSE otherwise

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);

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`id` (guint) –

stream ID where the sample should be sent

`sample` (GstSample *) –

RTP sample to send back

Returns (GstFlowReturn *) –
No description available

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);

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`id` (guint) –

stream ID where the sample should be sent

`sample` (GstSample *) –

RTP sample to send back

Returns (GstFlowReturn *) –
No description available

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.

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`parameter` (guint) –

the id of the stream for which to remove the key.

Returns (gboolean) –

TRUE when the command could be issued, FALSE otherwise

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.

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`parameter` (guint) –

a pointer to GstPromise

`arg1` (GstCaps *) –
No description available
`arg2` (GstPromise *) –
No description available
Returns (gboolean) –

TRUE when the command could be issued, FALSE otherwise

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.

Parameters:

`rtspsrc` (GstElement *) –

a rtspsrc

`parameter` (gchararray) –

a pointer to GstPromise

`arg1` (gchararray) –
No description available
`arg2` (gchararray) –
No description available
`arg3` (GstPromise *) –
No description available
Returns (gboolean) –

TRUE when the command could be issued, FALSE otherwise

Flags: Run Last / Action

## Properties

### 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 : 1.24.10

### 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

## Named constants

### Rtspbackchannel

#### Members

`none` (0) – No backchannel
`onvif` (1) – ONVIF audio backchannel

### Rtspnat-method

Different methods for trying to traverse firewalls.

#### Members

`none` (0) – None
`dummy` (1) – Send Dummy packets

### Rtspsrc-buffer-mode

#### Members

`none` (0) – Only use RTP timestamps
`slave` (1) – Slave receiver to sender clock
`buffer` (2) – Do low/high watermark buffering
`auto` (3) – Choose mode depending on stream live
`synced` (4) – Synchronized sender and receiver clocks

### Rtspsrc-ntp-time-source

#### Members

`ntp` (0) – NTP time based on realtime clock
`unix` (1) – UNIX time based on realtime clock
`running-time` (2) – Running time based on pipeline clock
`clock-time` (3) – Pipeline clock time

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