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