rtp: Move from website to docs

index.md

@@ -84,4 +84,4 @@ developers planing on or already using GStreamer for their applications:
 
 ## RTP and RTSP support in GStreamer
 
-<a href="/documentation/rtp.html">RTSP and RTP component overview</a>
+[RTSP and RTP component overview](rtp.md)

rtp.md

@@ -0,0 +1,91 @@
+# 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 to 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).
+

sitemap.txt

@@ -126,3 +126,4 @@ index.md
 			pwg-licensing-advisory.md
 	sdk-legal-information.md
 	splitup.md
+	rtp.md