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docs: update docs and comments
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Wim Taymans
Wim Taymans
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3 changed files with 130 additions and 49 deletions
174
docs/README
174
docs/README
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@ -103,7 +103,7 @@ can build simple server applications with it.
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alternative implementation can be used by the server.
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The GstRTSPMediaMapping object is more interesting and needs more configuration
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before the server object is useful. This object manages to mapping from a
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before the server object is useful. This object manages the mapping from a
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request URL to a specific stream and its configuration. We explain in the next
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topic how to configure this object.
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@ -202,7 +202,7 @@ can build simple server applications with it.
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have to be negotiated with the client in the SETUP requests.
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When preparing a GstRTSPMedia, a multifdsink is also constructed for streaming
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the stream over TCP^when requested.
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the stream over TCP when requested.
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* the GstRTSPClient object
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@ -212,61 +212,143 @@ can build simple server applications with it.
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a new GstRTCPClient object, will configure the session pool and media mapper
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objects in it and will then call the accept function of the client.
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The default GstRTSPClient will accept the connection and will start a new
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GThread to handle the connection. In RTSP it is usual to keep the connection
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open between multiple RTSP requests. The client thread will simply block for a
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new GstRTSPMessage, will dispatch it and will send a response.
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The default GstRTSPClient will accept the connection and will attach a watch to
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the server mainloop. In RTSP it is usual to keep the connection
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open between multiple RTSP requests. The client watch will be dispatched by the
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server mainloop when a new GstRTSPMessage is received, which will then be
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handled and a response will be sent.
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The GstRTSPClient object remains alive for as long as a client has a TCP
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connection open with the server. Since is possible for a client to open and close
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the TCP connection between requests, we cannot store the state related
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to the configured RTSP session in the GstRTSPClient object. This server state
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is instead stored in the GstRTSPSession object.
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* GstRTSPSession
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This object contains state about a specific RTSP session identified with a
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session id. This state contains the configured streams and their associated
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transports.
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We will briefly describe how it deals with some common requests.
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When a GstRTSPClient performs a SETUP request, the server will allocate a new
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GstRTSPSession with a unique session id from the GstRTSPSessionPool. The pool
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maintains a list of all existing sessions and makes sure that no session id is
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used multiple times. The session id is sent to the client so that the client
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can refer to its previously configured state by sending the session id in
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further requests.
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- DESCRIBE:
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A client will then use the session id to configure one or more streams,
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identified by their url. This information is kept in a GstRTSPSessionMedia
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structure that is refered to from the GstRTSPSession.
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locates the GstRTSPMedia for the url, prepares it and asks the sdp helper
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function to construct an SDP from the caps of the prepared media pipeline.
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It will also cache the url+media object so that it can be reused later.
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* GstRTSPSessionMedia and GstRTSPSessionStream
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A GstRTSPSessionMedia is identified by a URL and is referenced by a
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GstRTSPSession. It is created as soon as a client performs a SETUP operation on
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a particular URL. It will contain a link to the GstRTSPMedia object associated
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with the URL along with the state of the media and the configured transports
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for each of the streams in the media.
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Each SETUP request performed by the client will configure a
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GstRTSPSessionStream object linked to by the GstRTSPSessionMedia structure.
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It will contain the transport information needed to send this stream to the
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client. The GstRTSPSessionStream also contains a link to the GstRTSPMediaStream
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object that generates the actual data to be streamed to the client.
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Note how GstRTSPMedia and GstRTSPMediaStream (the providers of the data to
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stream) are decoupled from GstRTSPSessionMedia and GstRTSPSessionStream (the
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configuration of how to send this stream to a client) in order to be able to
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send the data of one GstRTSPMedia to multiple clients.
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* media control
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After a client has configured the transports for a GstRTSPMedia and its
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GstRTSPMediaStreams, the client can play/pause/stop the stream.
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The GstRTSPMedia object was prepared in the DESCRIBE call (or during SETUP when
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the client skipped the DESCRIBE request). As seen earlier, this configures a
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couple of multiudpsink and udpsrc elements to respectively send and receive the
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media to clients.
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When a client performs a PLAY request, its configured destination UDP ports are
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added to the GstRTSPMediaStream target destinations, at which point data will
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be sent to the client. The corresponding GstRTSPMedia object will be set to the
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PLAYING state if it was not allready in order to send the data to the
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destination.
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The server needs to prepare an RTP-Info header field in the PLAY response,
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which consists of the sequence number and the RTP timestamp of the next RTP
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packet. In order to achive this, the server queries the payloaders for this
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information when it prerolled the pipeline.
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When a client performs a PAUSE request, the destination UDP ports are removed
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from the GstRTSPMediaStream object and the GstRTSPMedia object is set to PAUSED
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if no other destinations are configured anymore.
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* seeking
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A seek is performed when a client sends a Range header in the PLAY request.
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This only works when not dealing with shared (live) streams.
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The server performs a GStreamer flushing seek on the media, waits for the
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pipeline to preroll again and then responds to the client after collecting the
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new RTP sequence number and timestamp from the payloaders.
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* session management
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The server has to react to clients that suddenly disappear because of network
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problems or otherwise. It needs to make sure that it can reasonable free the
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resources that are used by the various objects in use for streaming when the
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client appears to be gone.
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Each of the GstRTSPSession objects managed by a GstRTSPSessionPool has
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therefore a last_access field that contains the timestamp of when activity from
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a client was last recorded.
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- SETUP
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A new GstRTSPSession object will be created from the GstRTSPSessionPool
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object configured in the GstRTSPClient. This session will contain the
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configuration of the client regarding the media it is streaming and the
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ports/transport it negotiated with the server.
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The sessionid is set in the response header. The client will add the
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sessionid to any further SETUP/PLAY/PAUSE/TEARDOWN request so that we can
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always find the session again.
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The session configuration for a sessionid will have a link to the prepared
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GstRTSPMedia object of the stream. The port and transport of the client is
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stored in the session configuration.
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- PLAY
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The session configuration is retrieved with the sessionid and the client
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ports are configured in the UDP sinks, then the streaming to the client
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is started.
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- PAUSE
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The session configuration is retrieved with the sessionid and the client
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ports are removed from the UDP sinks, the streaming to the client
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pauses.
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- TEARDOWN
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The session configuration is released along with its link to the
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GstRTSPMedia object. When no more clients are refering to the GstRTSPMedia
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object, it can be released as well.
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Various ways exist to detect activity from a client:
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- RTSP keepalive requests. When a client is receiving RTP data, the RTSP TCP
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connection is largely unused. It is the client responsability to
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periodically send keep-alive requests over the TCP channel.
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Whenever a keep-alive request is received by the server (any request that
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contains a session id, usually an OPTION or GET_PARAMETER request) the
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last_access of the session is updated.
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- Since it is not required for a client to keep the RTSP TCP connection open
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while streaming, gst-rtsp-server also detects activity from clients by
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looking at the RTCP messages it receives.
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When an RTCP message is received from a client, the server looks in its list
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of active ports if this message originates from a known host/port pair that
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is currently active in a GstRTSPSession. If this is the case, the session is
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kept alive.
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Since the server does not know anything about the port number that will be
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used by the client to send RTCP, this method does not always work. Later
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RTSP RFCs will include support for negotiating this port number with the
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server. Most clients however use the same port number for sending and
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receiving RTCP exactly for this reason.
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If there was no activity in a particular session for a long time (by default 60
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seconds), the sessionpool will destroy the session along with all related
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objects and media as if a TEARDOWN happened from the client.
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* TEARDOWN
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A TEARDOWN request will first location the GstRTSPSessionMedia of the URL. It
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will then remove all transports from the streams, making sure that streaming
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stops to the client. It will then remove the GstRTSPSessionMedia and
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GstRTSPSessionStream structures. Finally the GstRTSPSession is released back
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into the pool.
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When there are no more references to the GstRTSPMedia, the media pipeline is
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shut down and destroyed.
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@ -1475,7 +1475,7 @@ static GstRTSPWatchFuncs watch_funcs = {
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* @client: a #GstRTSPClient
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* @channel: a #GIOChannel
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*
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* Accept a new connection for @client on the socket in @source.
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* Accept a new connection for @client on the socket in @channel.
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*
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* This function should be called when the client properties and urls are fully
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* configured and the client is ready to start.
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@ -464,8 +464,7 @@ default_accept_client (GstRTSPServer *server, GIOChannel *channel)
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/* set the session pool that this client should use */
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gst_rtsp_client_set_session_pool (client, server->session_pool);
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/* set the session pool that this client should use */
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/* set the media mapping that this client should use */
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gst_rtsp_client_set_media_mapping (client, server->media_mapping);
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/* accept connections for that client, this function returns after accepting
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