mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 05:31:15 +00:00
90 lines
5.4 KiB
Markdown
90 lines
5.4 KiB
Markdown
# RTP and RTSP support
|
|
|
|
GStreamer has excellent support for both RTP and RTSP, and its RTP/RTSP
|
|
stack has proved itself over years of being widely used in production use
|
|
in a variety of mission-critical and low-latency scenarios, from small
|
|
embedded devices to large-scale videoconferencing and command-and-control
|
|
systems.
|
|
|
|
## GStreamer RTSP Server
|
|
|
|
GStreamer's RTSP server (gst-rtsp-server) is a featureful and easy-to-use
|
|
library that allows applications to implement a complete RTSP server with
|
|
just a couple of lines of code.
|
|
|
|
It is multi-threaded, scalable and flexible, and provides support for
|
|
static or dynamic mount points, authentication, retransmission (rtx),
|
|
encryption (srtp, secure RTP), UDP unicast and multicast as well as
|
|
TCP interleaving, seeking, and optionally also cgroup integration for
|
|
advanced resource management and control. It can also distribute a
|
|
GStreamer net client clock to GStreamer RTSP clients to facilitate
|
|
multi-device synchronization.
|
|
|
|
|
|
## GStreamer RTSP Client
|
|
|
|
The GStreamer <tt>rtspsrc</tt> element from gst-plugins-good is GStreamer's
|
|
high-level RTSP client abstraction. It can be used as a standalone element
|
|
directly, or can be used via <tt>playbin</tt> by passing an rtsp:// URI to
|
|
playbin. <tt>rtspsrc</tt> features a number of GObject properties that allow
|
|
you to configure it in all kinds of different ways, most notably a
|
|
<tt>"latency"</tt> property to configure the default jitterbuffer latency,
|
|
which you may want to configure to a lower value to achieve lower latency.
|
|
|
|
|
|
## RTP components
|
|
|
|
Most of GStreamer's key RTP components live in gst-plugins-good:
|
|
* The <tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-plugin-rtpmanager.html">rtpmanager</a></tt>
|
|
plugin contains elements like
|
|
<tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpbin.html">rtpbin</a></tt> and
|
|
<tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpjitterbuffer.html">rtpjitterbuffer</a></tt>
|
|
* The <tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-plugin-rtp.html">rtp</a></tt> plugin
|
|
contains RTP payloading and depayloading elements for many different
|
|
codecs and container formats
|
|
|
|
with some lower-level libraries in gst-plugins-base:
|
|
|
|
* The <a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gstreamer-rtp.html">GStreamer RTP library</a>
|
|
contains things such as RTP payloader/depayloader base classes and functions to handle RTP and RTCP buffers
|
|
* The <a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gstreamer-mikey.html">GStreamer MIKEY library</a>
|
|
contains helper functions to deal with MIKEY messages for secure RTP
|
|
* The <a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gstreamer-rtsp.html">GStreamer RTSP library</a>
|
|
contains low-level RTSP functionality used by gst-rtsp-server and higher-level objects such as rtspsrc.
|
|
* The <a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gstreamer-sdp.html">GStreamer SDP library</a>
|
|
contains utility functions for SDP message parsing and creation.
|
|
|
|
|
|
Some of the main components are:
|
|
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpbin.html">rtpbin</a></tt>
|
|
is the high-level RTP component and supports sending
|
|
and receiving, just sending or just receiving data, with and without RTCP
|
|
support. This is the bin that does it all: it adapts dynamically to your
|
|
needs based on the requested pads; it also contains an rtpjitterbuffer.
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpjitterbuffer.html">rtpjitterbuffer</a></tt>
|
|
is an RTP buffer that controls network jitter and reorders packets. It also
|
|
dumps packets that arrive too late, handles packet retransmission and lost
|
|
packet notification and adjusts for sender-receiver clock drift.
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpptdemux.html">rtpptdemux</a></tt>
|
|
is an element that usually sits on the rtpbin src
|
|
pad and will detect any new payload types that arrive in the RTP stream.
|
|
It will then create a pad for that new payload and you can connect a
|
|
depayloader/decoder pipeline to that pad.
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-good-plugins/html/gst-plugins-good-plugins-rtpssrcdemux.html">rtpssrcdemux</a></tt>
|
|
is an element that usually sits on the rtpbin src
|
|
pad and will detect any new SSRCs that arrive in the RTP stream.
|
|
It will then create a pad for that new SSRC and you can connect a
|
|
depayloader/decoder pipeline to that pad.
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtpbasedepayload.html">GstRTPBaseDepayload</a></tt>
|
|
is a base class for RTP depayloaders
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtpbasepayload.html">GstRTPBasePayload</a></tt>
|
|
is a base class for RTP payloaders
|
|
* <tt><a href="/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtpbaseaudiopayload.html">GstRTPBaseAudioPayload</a>
|
|
is a base class for audio RTP payloaders
|
|
|
|
|
|
|
|
Note that many RTP elements assume they receive RTP buffers with
|
|
<a href="/data/doc/gstreamer/head/gstreamer-libs/html/gstreamer-libs-GstNetAddressMeta.html">GstNetAddressMeta</a>
|
|
meta data set on them (as udpsrc will produce).
|