rtspsrc

rtspsrc — Receive data over the network via RTSP (RFC 2326)

Synopsis




                    GstRTSPSrc;

Object Hierarchy


  GObject
   +----GstObject
         +----GstOsxAudioSink
               +----GstRTSPSrc

Properties


  "debug"                    gboolean              : Read / Write / Construct
  "location"                 gchararray            : Read / Write
  "protocols"                GstRTSPLowerTrans     : Read / Write / Construct
  "retry"                    guint                 : Read / Write / Construct
  "timeout"                  guint64               : Read / Write / Construct
  "latency"                  guint                 : Read / Write / Construct
  "tcp-timeout"              guint64               : Read / Write / Construct

Description

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_streamd 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 however currently not yet implemented.

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

Example launch line

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

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

Last reviewed on 2006-08-18 (0.10.5)

Element Information

plugin rtsp
author Wim Taymans <wim@fluendo.com> Thijs Vermeir <thijs.vermeir@barco.com> Lutz Mueller <lutz@topfrose.de>
class Source/Network

Details

GstRTSPSrc

typedef struct _GstRTSPSrc GstRTSPSrc;

Property Details

The "debug" property

  "debug"                    gboolean              : Read / Write / Construct

Dump request and response messages to stdout.

Default value: FALSE

The "location" property

  "location"                 gchararray            : Read / Write

Location of the RTSP url to read.

Default value: NULL

The "protocols" property

  "protocols"                GstRTSPLowerTrans     : Read / Write / Construct

Allowed lower transport protocols.

Default value: UDP Unicast Mode|UDP Multicast Mode|TCP interleaved mode

The "retry" property

  "retry"                    guint                 : Read / Write / Construct

Max number of retries when allocating RTP ports.

Allowed values: <= 65535

Default value: 20

The "timeout" property

  "timeout"                  guint64               : Read / Write / Construct

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

Default value: 5000000

The "latency" property

  "latency"                  guint                 : Read / Write / Construct

Amount of ms to buffer.

Default value: 3000

The "tcp-timeout" property

  "tcp-timeout"              guint64               : Read / Write / Construct

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

Default value: 20000000