gstreamer/docs

README 
------

(Last updated on Fri 30 jan 2009, version 0.10.1.1)

This HOWTO describes the basic usage of the GStreamer RTSP libraries and how you
can build simple server applications with it.

* General

 The server relies heavily on the RTSP infrastructure of GStreamer. This includes
 all of the media acquisition, decoding, encoding, payloading and UDP/TCP
 streaming. We use the gstrtpbin element for all the session management. Most of
 the RTSP message parsing and construction in the server is done using the RTSP
 library that comes with gst-plugins-base. 

 The result is that the server is rather small (a few 1000 lines of code) and easy
 to understand and extend. In its current state of development, things change
 fast, API and ABI are unstable. We encourage people to use it for their various
 use cases and participate by suggesting changes/features.

 Most of the server is built as a library containing a bunch of GObject objects
 that provide reasonable default functionality but has a fair amount of hooks
 to override the default behaviour.

 The server currently integrates with the glib mainloop nicely. It is also a
 heavy user of multiple threads. It's currently not meant to be used in
 high-load scenarios and you should probably not put it on a public IP address.

* Initialisation

 You need to initialize GStreamer before using any of the RTSP server functions.

   #include <gst/gst.h>

   int
   main (int argc, char *argv[])
   {
     gst_init (&argc, &argv);

     ...
   }
 
 The server itself currently does not have any specific initialisation function
 but that might change in the future.


* Creating the server

 The first thing you want to do is create a new GstRTSPServer object. This object
 will handle all the new client connections to your server once it is added to a
 GMainLoop. You can create a new server object like this:

   #include <gst/rtsp-server/rtsp-server.h>

   GstRTSPServer *server;

   server = gst_rtsp_server_new ();

 The server will by default listen on port 8554 for new connections. This can be
 changed by calling gst_rtsp_server_set_port() or with the 'port' GObject
 property. This makes it possible to run multiple server instances listening on
 multiple ports on one machine.

 We can make the server start listening on its default port by attaching it to a
 mainloop. The following example shows how this is done and will start a server
 on the default 8554 port. For any request we make, we will get a NOT_FOUND
 error code because we need to configure more things before the server becomes
 useful.

   #include <gst/gst.h>
   #include <gst/rtsp-server/rtsp-server.h>

   int
   main (int argc, char *argv[])
   {
     GstRTSPServer *server;
     GMainLoop *loop;

     gst_init (&argc, &argv);

     server = gst_rtsp_server_new ();

     /* make a mainloop for the default context */
     loop = g_main_loop_new (NULL, FALSE);

     /* attach the server to the default maincontext */
     gst_rtsp_server_attach (server, NULL);

     /* start serving */
     g_main_loop_run (loop);
   }

 The server manages two other objects: GstRTSPSessionPool and
 GstRTSPMediaMapping. 

 The GstRTSPSessionPool is an object that keeps track of all the active sessions
 in the server. A session will usually be kept for each client that performed a
 SETUP request for a certain media stream. It contains the configuration that
 the client negotiated with the server to receive the particular stream, ie. the
 transport used and port pairs for UDP along with the state of the streaming.
 The default implementation of the session pool is usually sufficient but
 alternative implementation can be used by the server.

 The GstRTSPMediaMapping object is more interesting and needs more configuration
 before the server object is useful. This object manages to mapping from a
 request URL to a specific stream and its configuration. We explain in the next
 topic how to configure this object.


* Making url mappings

 Next we need to define what media is attached to a particular URL. What we want
 to achieve is that when the user asks our server for a specific URL, say /test,
 that we create (or reuse) a GStreamer pipeline that produces one or more RTP
 streams. 
 
 The object that can create such pipeline is called a GstRTSPMediaFactory object. 
 The default implementation of GstRTSPMediaFactory allows you to easily create
 GStreamer pipelines using the gst-launch syntax. It possible to create a
 GstRTSPMediaFactory subclass that uses different methods for constructing
 pipelines.

 The default GstRTSPMediaFactory can be configured with a gst-launch line that
 produces a toplevel bin (use '(' and ')' around the pipeline description to
 force a toplevel GstBin instead of the default GstPipeline toplevel element).
 The pipeline description should contain elements named payN, one for each
 stream (ex. pay0, pay1, ...). Also, for increased compatibility each stream
 should have a different payload type which can be configured on the payloader.

 The following code snippet illustrates how to create a media factory that
 creates an RTP feed of an H264 encoded test video signal.

   GstRTSPMediaFactory *factory;

   factory = gst_rtsp_media_factory_new ();

   gst_rtsp_media_factory_set_launch (factory, 
                "( videotestsrc ! x264enc ! rtph264pay pt=96 name=pay0 )");

 Now that we have the media factory, we can attach it to a specific url. To do
 this we get the default GstRTSPMediaMapping from our server and add the url to
 factory mapping to it like this:

   GstRTSPMediaMapping *mapping;

   ...create server..create factory..

   /* get the default mapping from the server */
   mapping = gst_rtsp_server_get_media_mapping (server);

   /* attach the video test signal to the "/test" URL */
   gst_rtsp_media_mapping_add_factory (mapping, "/test", factory);
   g_object_unref (mapping);

 When starting the server now and directing an RTP client to the URL (like with 
 vlc, mplayer or gstreamer):

   rtsp://localhost:8554/test 

 a test signal will be streamed to the client. The full example code can be
 found in the examples/test-readme.c file.


* more on GstRTSPMediaFactory

 The GstRTSPMediaFactory is responsible for creating and caching GstRTSPMedia
 objects. 

 A freshly created GstRTSPMedia object from the factory initialy only contains a
 GstElement containing the elements to produce the RTP streams for the media and
 a GArray of GstRTSPMediaStream objects describing the payloader and its source
 pad. The media is unprepared in this state.

 Usually the url will determine what kind of pipeline should be created. You can
 for example use query parameters to configure certain parts of the pipeline or
 select encoders and payloaders based on some url pattern.

 When dealing with a live stream from, for example, a webcam, it can be
 interesting to share the pipeline with multiple clients. This must be done when
 only one instance of the video capture element can be used at a time. In this
 case, the shared property of GstRTSPMedia must be used to instruct the default
 GstRTSPMediaFactory implementation to cache the media.

 When all objects created from a factory can be shared, you can set the shared
 property directly on the factory.

* more on GstRTSPMedia

 After creating the GstRTSPMedia object from the factory, it can be prepared
 with gst_rtsp_media_prepare(). This method will put those objects in a
 GstPipeline and will construct and link the streaming elements and the
 gstrtpbin session manager object.

 The _prepare() method will then preroll the pipeline in order to figure out the
 caps on the payloaders. After the GstRTSPMedia prerolled it will be in the
 prepared state and can be used for creating SDP files or for streaming to
 clients.

 The prepare method will also create 2 UDP ports for each stream that can be
 used for sending and receiving RTP/RTCP from clients. These port numbers will
 have to be negotiated with the client in the SETUP requests.

 When preparing a GstRTSPMedia, a multifdsink is also constructed for streaming
 the stream over TCP^when requested.


* the GstRTSPClient object

 When a server detects a new client connection on its port, it will call its
 accept_client vmethod. The default implementation of this function will create
 a new GstRTCPClient object, will configure the session pool and media mapper
 objects in it and will then call the accept function of the client.

 The default GstRTSPClient will accept the connection and will start a new
 GThread to handle the connection. In RTSP it is usual to keep the connection
 open between multiple RTSP requests. The client thread will simply block for a
 new GstRTSPMessage, will dispatch it and will send a response.
 
 We will briefly describe how it deals with some common requests.

 - DESCRIBE:

    locates the GstRTSPMedia for the url, prepares it and asks the sdp helper
    function to construct an SDP from the caps of the prepared media pipeline.
    It will also cache the url+media object so that it can be reused later.
 
 - SETUP

    A new GstRTSPSession object will be created from the GstRTSPSessionPool
    object configured in the GstRTSPClient. This session will contain the
    configuration of the client regarding the media it is streaming and the
    ports/transport it negotiated with the server.

    The sessionid is set in the response header. The client will add the
    sessionid to any further SETUP/PLAY/PAUSE/TEARDOWN request so that we can
    always find the session again.

    The session configuration for a sessionid will have a link to the prepared
    GstRTSPMedia object of the stream. The port and transport of the client is
    stored in the session configuration.

 - PLAY

    The session configuration is retrieved with the sessionid and the client
    ports are configured in the UDP sinks, then the streaming to the client 
    is started. 

 - PAUSE

    The session configuration is retrieved with the sessionid and the client
    ports are removed from the UDP sinks, the streaming to the client 
    pauses. 

 - TEARDOWN

    The session configuration is released along with its link to the
    GstRTSPMedia object. When no more clients are refering to the GstRTSPMedia
    object, it can be released as well.