refer to the linux referen manual, the version is 3.14.52
RTSP streams can be played with a manual pipline or by using playbin. The format of the commands is as follows.
• Manual pipeline
gst-launch-1.0 rtspsrc location=$RTSP_URI name=source
! queue ! $video_rtp_depacketize_plugin ! $vpu_dec ! $video_sink_plugin source.
! queue ! $audio_rtp_depacketize_plugin ! $audio_parse_plugin ! beepdec !
$audio_sink_plugin
Two properties of rtspsrc that are useful for RTSP streaming are:
• Latency: This is the extra added latency of the pipeline, with the default value of 200 ms. If you need low-latency
RTSP streaming playback, you can set this property to a smaller value.
• Buffer-mode: This property is used to control the buffering algorithm in use. It includes four modes:
• None: Outgoing timestamps are calculated directly from the RTP timestamps, not good for real-time applications.
• Slave: Calculates the skew between the sender and receiver and produces smoothed adjusted outgoing
timestamps, good for low latency communications.
• Buffer: Buffer packets between low and high watermarks, good for streaming communication.
• Auto: Chooses the three modes above depending on the stream. This is the default setting.
If you need to pause or resume the RTSP streaming playback, you need to use a buffer-mode of slave or none for rtspsrc,
as in buffer-mode=buffer. After resuming, the timestamp is forced to start from 0, and this will cause buffers to be
dropped after resuming.
Manual pipeline example:
gst-launch-1.0 rtspsrc location=rtsp://10.192.241.11:8554/test name=source
! queue ! rtph264depay ! vpudec ! overlaysink source.
! queue ! rtpmp4gdepay ! aacparse ! beepdec ! pulsesink
Playback will not exit automatically in GStreamer 1.x, if buffer-mode is set to buffer in the rtpsrc plugin.
