docs: update docs and comments

This commit is contained in:
Wim Taymans 2009-12-25 18:24:10 +01:00 committed by Wim Taymans
parent 92eb244215
commit 996112db95
3 changed files with 130 additions and 49 deletions

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@ -103,7 +103,7 @@ can build simple server applications with it.
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
before the server object is useful. This object manages the mapping from a
request URL to a specific stream and its configuration. We explain in the next
topic how to configure this object.
@ -202,7 +202,7 @@ can build simple server applications with it.
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 stream over TCP when requested.
* the GstRTSPClient object
@ -212,61 +212,143 @@ can build simple server applications with it.
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.
The default GstRTSPClient will accept the connection and will attach a watch to
the server mainloop. In RTSP it is usual to keep the connection
open between multiple RTSP requests. The client watch will be dispatched by the
server mainloop when a new GstRTSPMessage is received, which will then be
handled and a response will be sent.
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.
The GstRTSPClient object remains alive for as long as a client has a TCP
connection open with the server. Since is possible for a client to open and close
the TCP connection between requests, we cannot store the state related
to the configured RTSP session in the GstRTSPClient object. This server state
is instead stored in the GstRTSPSession object.
* GstRTSPSession
This object contains state about a specific RTSP session identified with a
session id. This state contains the configured streams and their associated
transports.
When a GstRTSPClient performs a SETUP request, the server will allocate a new
GstRTSPSession with a unique session id from the GstRTSPSessionPool. The pool
maintains a list of all existing sessions and makes sure that no session id is
used multiple times. The session id is sent to the client so that the client
can refer to its previously configured state by sending the session id in
further requests.
A client will then use the session id to configure one or more streams,
identified by their url. This information is kept in a GstRTSPSessionMedia
structure that is refered to from the GstRTSPSession.
* GstRTSPSessionMedia and GstRTSPSessionStream
A GstRTSPSessionMedia is identified by a URL and is referenced by a
GstRTSPSession. It is created as soon as a client performs a SETUP operation on
a particular URL. It will contain a link to the GstRTSPMedia object associated
with the URL along with the state of the media and the configured transports
for each of the streams in the media.
Each SETUP request performed by the client will configure a
GstRTSPSessionStream object linked to by the GstRTSPSessionMedia structure.
It will contain the transport information needed to send this stream to the
client. The GstRTSPSessionStream also contains a link to the GstRTSPMediaStream
object that generates the actual data to be streamed to the client.
Note how GstRTSPMedia and GstRTSPMediaStream (the providers of the data to
stream) are decoupled from GstRTSPSessionMedia and GstRTSPSessionStream (the
configuration of how to send this stream to a client) in order to be able to
send the data of one GstRTSPMedia to multiple clients.
* media control
After a client has configured the transports for a GstRTSPMedia and its
GstRTSPMediaStreams, the client can play/pause/stop the stream.
The GstRTSPMedia object was prepared in the DESCRIBE call (or during SETUP when
the client skipped the DESCRIBE request). As seen earlier, this configures a
couple of multiudpsink and udpsrc elements to respectively send and receive the
media to clients.
When a client performs a PLAY request, its configured destination UDP ports are
added to the GstRTSPMediaStream target destinations, at which point data will
be sent to the client. The corresponding GstRTSPMedia object will be set to the
PLAYING state if it was not allready in order to send the data to the
destination.
The server needs to prepare an RTP-Info header field in the PLAY response,
which consists of the sequence number and the RTP timestamp of the next RTP
packet. In order to achive this, the server queries the payloaders for this
information when it prerolled the pipeline.
When a client performs a PAUSE request, the destination UDP ports are removed
from the GstRTSPMediaStream object and the GstRTSPMedia object is set to PAUSED
if no other destinations are configured anymore.
* seeking
A seek is performed when a client sends a Range header in the PLAY request.
This only works when not dealing with shared (live) streams.
The server performs a GStreamer flushing seek on the media, waits for the
pipeline to preroll again and then responds to the client after collecting the
new RTP sequence number and timestamp from the payloaders.
* session management
The server has to react to clients that suddenly disappear because of network
problems or otherwise. It needs to make sure that it can reasonable free the
resources that are used by the various objects in use for streaming when the
client appears to be gone.
Each of the GstRTSPSession objects managed by a GstRTSPSessionPool has
therefore a last_access field that contains the timestamp of when activity from
a client was last recorded.
Various ways exist to detect activity from a client:
- RTSP keepalive requests. When a client is receiving RTP data, the RTSP TCP
connection is largely unused. It is the client responsability to
periodically send keep-alive requests over the TCP channel.
Whenever a keep-alive request is received by the server (any request that
contains a session id, usually an OPTION or GET_PARAMETER request) the
last_access of the session is updated.
- Since it is not required for a client to keep the RTSP TCP connection open
while streaming, gst-rtsp-server also detects activity from clients by
looking at the RTCP messages it receives.
When an RTCP message is received from a client, the server looks in its list
of active ports if this message originates from a known host/port pair that
is currently active in a GstRTSPSession. If this is the case, the session is
kept alive.
Since the server does not know anything about the port number that will be
used by the client to send RTCP, this method does not always work. Later
RTSP RFCs will include support for negotiating this port number with the
server. Most clients however use the same port number for sending and
receiving RTCP exactly for this reason.
If there was no activity in a particular session for a long time (by default 60
seconds), the sessionpool will destroy the session along with all related
objects and media as if a TEARDOWN happened from the client.
* TEARDOWN
A TEARDOWN request will first location the GstRTSPSessionMedia of the URL. It
will then remove all transports from the streams, making sure that streaming
stops to the client. It will then remove the GstRTSPSessionMedia and
GstRTSPSessionStream structures. Finally the GstRTSPSession is released back
into the pool.
When there are no more references to the GstRTSPMedia, the media pipeline is
shut down and destroyed.

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@ -1475,7 +1475,7 @@ static GstRTSPWatchFuncs watch_funcs = {
* @client: a #GstRTSPClient
* @channel: a #GIOChannel
*
* Accept a new connection for @client on the socket in @source.
* Accept a new connection for @client on the socket in @channel.
*
* This function should be called when the client properties and urls are fully
* configured and the client is ready to start.

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@ -464,8 +464,7 @@ default_accept_client (GstRTSPServer *server, GIOChannel *channel)
/* set the session pool that this client should use */
gst_rtsp_client_set_session_pool (client, server->session_pool);
/* set the session pool that this client should use */
/* set the media mapping that this client should use */
gst_rtsp_client_set_media_mapping (client, server->media_mapping);
/* accept connections for that client, this function returns after accepting