GStreamer Good Plugins 0.10 Plugins Reference Manual | ||||
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Top | Description | Object Hierarchy | Implemented Interfaces | Properties |
"debug" gboolean : Read / Write "location" gchar* : Read / Write "protocols" GstRTSPLowerTrans : Read / Write "retry" guint : Read / Write "timeout" guint64 : Read / Write "latency" guint : Read / Write "tcp-timeout" guint64 : Read / Write "connection-speed" guint : Read / Write "nat-method" GstRTSPNatMethod : Read / Write "do-rtcp" gboolean : Read / Write "proxy" gchar* : Read / Write "rtp-blocksize" guint : Read / Write "user-id" gchar* : Read / Write "user-pw" gchar* : Read / Write
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 currently fully implemented with the gstrtpbin in the gst-plugins-bad module.
rtspsrc acts like a live source and will therefore only generate data in the PLAYING state.
gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
Last reviewed on 2006-08-18 (0.10.5)
"debug"
property "debug" gboolean : Read / Write
Dump request and response messages to stdout.
Default value: FALSE
"location"
property "location" gchar* : Read / Write
Location of the RTSP url to read.
Default value: NULL
"protocols"
property "protocols" GstRTSPLowerTrans : Read / Write
Allowed lower transport protocols.
Default value: UDP Unicast Mode|UDP Multicast Mode|TCP interleaved mode
"retry"
property "retry" guint : Read / Write
Max number of retries when allocating RTP ports.
Allowed values: <= 65535
Default value: 20
"timeout"
property "timeout" guint64 : Read / Write
Retry TCP transport after UDP timeout microseconds (0 = disabled).
Default value: 5000000
"tcp-timeout"
property "tcp-timeout" guint64 : Read / Write
Fail after timeout microseconds on TCP connections (0 = disabled).
Default value: 20000000
"connection-speed"
property "connection-speed" guint : Read / Write
Network connection speed in kbps (0 = unknown).
Allowed values: <= 2147483
Default value: 0
"nat-method"
property "nat-method" GstRTSPNatMethod : Read / Write
Method to use for traversing firewalls and NAT.
Default value: Send Dummy packets
"do-rtcp"
property "do-rtcp" gboolean : Read / Write
Send RTCP packets, disable for old incompatible server.
Default value: TRUE
"proxy"
property "proxy" gchar* : Read / Write
Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port].
Default value: NULL
"rtp-blocksize"
property "rtp-blocksize" guint : Read / Write
RTP package size to suggest to server (0 = disabled).
Allowed values: <= 65536
Default value: 0
"user-id"
property "user-id" gchar* : Read / Write
RTSP location URI user id for authentication.
Default value: NULL