[gst-rtsp-server] set ports

[Paulo Paiva]

Hello!

Is it possible to configure the negotiated rtp ports used to 
send/receive/control audio/video in the gstrtspserver.c?
I have been reading the documentation and in the GstRTSPMediaTrans class 
there is another class GstRTSPTransport with the fields GstRTSPRange   
server_port; I suppose i can define here the port range for the server.

I have created the gstrtspmedia and set the min and max ports, but now 
how do i tell the factory to use that configuration (if this is even 
correct) code bellow.
(Section added

     media =  gst_rtsp_media_new();
     mediastream = gst_rtsp_media_get_stream(media,idx);
     mediastream->server_port.min=3007;
     mediastream->server_port.max=3030;

)

Thanks all and regards!

-- 
Paulo Paiva

/* GStreamer
  * Copyright (C) 2008 Wim Taymans<wim.taymans at gmail.com>
  *
  * This library is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Library General Public
  * License as published by the Free Software Foundation; either
  * version 2 of the License, or (at your option) any later version.
  *
  * This library is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Library General Public License for more details.
  *
  * You should have received a copy of the GNU Library General Public
  * License along with this library; if not, write to the
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  *
  * gst-launch playbin uri=rtsp://127.0.0.1:3001/fileplay
  *
  */

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

/* define this if you want the resource to only be available when using
  * user/admin as the password */
#undef WITH_AUTH

char *filein = "movie.mkv";

/* this timeout is periodically run to clean up the expired sessions from the
  * pool. This needs to be run explicitly currently but might be done
  * automatically as part of the mainloop. */
static gboolean
timeout (GstRTSPServer * server, gboolean ignored)
{
   GstRTSPSessionPool *pool;

   pool = gst_rtsp_server_get_session_pool (server);
   gst_rtsp_session_pool_cleanup (pool);
   g_object_unref (pool);

   return TRUE;
}

int
main (int argc, char *argv[])
{
   GMainLoop *loop;
   GstRTSPServer *server;
   GstRTSPMediaMapping *mapping;
   GstRTSPMediaFactory *factory;
     GstRTSPMedia *media;
     GstRTSPMediaStream *mediastream;
     guint idx;
#ifdef WITH_AUTH
   GstRTSPAuth *auth;
   gchar *basic;
#endif
   gchar *str;

   gst_init (&argc,&argv);

   loop = g_main_loop_new (NULL, FALSE);

   /* create a server instance */
   server = gst_rtsp_server_new ();

   /* set server address */
   /* gst_rtsp_server_set_address(server, "0.0.0.0");*/

   /* set server listening port*/
   gst_rtsp_server_set_port(server, 3001);

   /* get the mapping for this server, every server has a default mapper object
    * that be used to map uri mount points to media factories */
   mapping = gst_rtsp_server_get_media_mapping (server);

#ifdef WITH_AUTH
   /* make a new authentication manager. it can be added to control access to all
    * the factories on the server or on individual factories. */
   auth = gst_rtsp_auth_new ();
   basic = gst_rtsp_auth_make_basic ("user", "admin");
   gst_rtsp_auth_set_basic (auth, basic);
   g_free (basic);
   /* configure in the server */
   gst_rtsp_server_set_auth (server, auth);
#endif

   /* make the play string*/
   str = g_strdup_printf ("( "
       "filesrc location=%s ! 'decodebin name=demux ,width=(int)352, height=(int)288' ! queue ! ffmpegcolorspace !"
       "x264enc tune=zerolatency byte-stream=true bitrate=1050 threads=0 speed-preset=3 ! rtph264pay name=pay0 pt=96 "
       "demux. ! queue ! audioresample ! audioconvert ! ffenc_aac ! rtpmp4apay name=pay1 pt=97 " ")",filein);

   /* make a media factory for a test stream. The default media factory can use
    * gst-launch syntax to create pipelines.
    * any launch line works as long as it contains elements named pay%d. Each
    * element with pay%d names will be a stream */
   factory = gst_rtsp_media_factory_new ();
     media =  gst_rtsp_media_new();
     mediastream = gst_rtsp_media_get_stream(media,idx);
     mediastream->server_port.min=3007;
     mediastream->server_port.max=3030;
   gst_rtsp_media_factory_set_launch (factory, str);

   /* uncoment in order to share the content in the pipe */
   /*gst_rtsp_media_factory_set_shared()*/

   /* attach the test factory to the /fileplay url */
   gst_rtsp_media_mapping_add_factory (mapping, "/fileplay", factory);

   /* don't need the ref to the mapper anymore */
   g_object_unref (mapping);

   /* attach the server to the default maincontext */
   if (gst_rtsp_server_attach (server, NULL) == 0)
     goto failed;

   /* add a timeout for the session cleanup */
   g_timeout_add_seconds (2, (GSourceFunc) timeout, server);

   /* start serving, this never stops */
   g_main_loop_run (loop);

   return 0;

   /* ERRORS */
failed:
   {
     g_print ("failed to attach the server\n");
     return -1;
   }
}